CN101641970B - A method and encoder for combining digital data sets, a decoding method and decoder for such combined digital data sets and a record carrier for storing such combined digital data set - Google Patents
A method and encoder for combining digital data sets, a decoding method and decoder for such combined digital data sets and a record carrier for storing such combined digital data set Download PDFInfo
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Abstract
Two digital data sets are combined by equating a first subset of samples to neighboring samples from a second subset which is interleaved with the first subset of samples where the equated samples of the two digital data sets do not correspond in time, and by subsequently adding corresponding samples from both digital data sets. This results in a third digital data set that allows the unraveling of the two digital data sets. The third digital data set, when combining two digital audio streams into a single digital audio stream, is still a good mono representation of the two combined digital audio streams and can thus be reproduced on regular reproduction equipment, yet the use of a decoder according to the invention allows the unraveling of the two digital data sets from the third digital data set.
Description
Technical field
The present invention relates to a kind of method; The second sample digital data sets that is used for will having the first sample digital data sets of first size and having second size is combined into the 3rd sample digital data sets with the 3rd size, and the 3rd size is less than this first size and this second size sum.
Background technology
From the known this method of EP1592008, disclosed a kind of method that is used for two digital data sets are mixed into the 3rd digital data sets therein.In order two digital data sets to be fitted to the single digital data sets that has less than the size of the size sum of these two digital data sets, the information that needs these two numerical datas of reduction to concentrate.Insert through concentrating in first numerical data to concentrate in the incomparable inconsistent one group of sample place definition between the precalculated position at the sample place between first group of precalculated position with in second numerical data, EP1592008 has realized that this information reduces.In being set as, inserts sample numerical value (value) between the precalculated position of digital data sets numerical value.Be to merge each sample of (total) first digital data sets and the respective sample of second digital data sets after this information reduction of these two numerical data concentrative implementation.This has produced the 3rd digital data sets that comprises this merging sample.Though only be utilized between the precalculated position interior reconstitute this, be utilized in known offset relation between the precalculated position between first digital data sets and second digital data sets, this sample is merged together allow recovery first digital data sets and second digital data sets.When the method for EP1592008 is used to audio stream, insert in this and not obvious and can play the 3rd digital data sets as the hybrid representation of two included digital data sets.In order to make it possible to utilize interior reconstitute originally to recover first and second digital data sets; Must know and be used for the two initial value of first and second digital data sets; And therefore during mixing these two numerical value also are stored, to allow separating these two digital data sets from the 3rd digital data sets afterwards.
The method of EP1592008 has following shortcoming, that is, it requires on coding one side, to focus on.
Summary of the invention
The objective of the invention is to reduce the processing that on the coding side, requires.
In order to achieve this end, method of the present invention may further comprise the steps:
-the first sample subclass of first digital data sets etc. is changed into the adjacent sample in the second sample subclass of (equate) first digital data sets, wherein this first sample subclass is staggered with this second sample subclass,
-the 3rd sample subclass of second digital data sets etc. is changed into the adjacent sample in the 4th sample subclass of second digital data sets, wherein the 3rd sample subclass is staggered with the 4th sample subclass,
-through in time domain, the sample of first digital data sets and the corresponding sample addition of second digital data sets being produced the sample of the 3rd digital data sets,
-concentrate first seed specimen of this first digital data sets of embedding and second seed specimen of this second digital data sets in the 3rd numerical data.
Through utilizing following steps to substitute the interior slotting step in the method for EP1592008, that is, the numerical value between the precalculated position is set as the numerical value of adjacent sample, in the processing intensity quilt reduction widely of coding side.Consequential signal still allows to separate (promptly extracting) these two digital data sets from the 3rd digital data sets.When two digital audio streams were combined into single digital audio stream, the 3rd digital data sets remained good single-tone (mono) expression of these two digital audio streams that have been combined.
The present invention is based on following understanding; Promptly; It is unnecessary inserting on the coding side, this be because, because the sample that this combination and separation method keep first and second digital data sets is intact and can be resumed in their other pre-positions of branch; Therefore allow the sample interpolation between this intact sample after the decoding of the 3rd digital data sets, so insert in can on the decoding side, carrying out equally well.The 3rd digital data sets part that the 3rd digital data sets of independent claims of the present invention is different from EP1592008 is; In situation of the present invention, between the real merging of first and second digital data sets and the 3rd digital data sets, there is bigger error usually.
First sample subclass of first digital data sets etc. is changed into the adjacent sample in the second sample subclass of first digital data sets; Wherein the first sample subclass and the second sample subclass are staggered, in the information of first digital data sets, have realized the reduction that is performed easily.
The 3rd sample subclass of second digital data sets etc. is changed into the adjacent sample in the 4th sample subclass of second digital data sets; Wherein the 3rd sample subclass and the 4th sample subclass are staggered, in the information of second digital data sets, have realized the reduction that is performed easily.
Can get through feasible initial value from first and second digital data sets; Wherein this initial value can be used as seed numerical value; And guarantee that the second and the 4th subclass also interlocks; Therein the first sample subclass of first digital data sets the 3rd sample subclass that waited adjacent sample and second digital data sets in the second sample subclass change into first digital data sets by etc. change in the state of the adjacent sample in the 4th sample subclass of second digital data sets, can recover first and second digital data sets from the 3rd digital data sets.In case first and second digital data sets are resumed in this state, insert in just can using or filtering recovers the initial value of the 3rd sample subclass of the first sample subclass and second digital data stream of first digital data stream as far as possible exactly.Therefore this method that first digital data stream and second digital data stream is combined into the 3rd digital data stream allows to recover the second and the 4th sample subclass and reconstruct first and third value subclass with high precision; And during decoding; If desired, can carry out the interior step of inserting.
The end user device that comprises decoder can determine the quality level that this reconstruct realizes, this is because can be by inserting in decoder selection and the execution rather than being stipulated by encoder.
Through first and second digital data sets not being carried out inserting in any but to be included in the error of hiding in the least significant bit of the 3rd digital data stream approximate, realized following advantage, that is, decoding step can freely be selected which type of reconstruct of application.Yet; When also use error is approximate during making up the 3rd set of digits (being the sample mixing that comprises the 1st and the 2nd set of digits of approximate error); Thereby also must use the error numerical approximation execution initial number data set of in least significant bit, hiding, the i.e. reconstruct of initial number voice-grade channel during the decode procedure.
Reconstruct during decoding can be selected approximate and be between the sample number value in the precalculated position and carry out linear interpolation with the error using as in least significant bit, store; This be because; Except the information loss in least significant bit, they can be resumed fully.Therefore can use the Code And Decode system more neatly.
Coding can or only reduce be handled and first and second digital data streams are fused into the 3rd digital data stream and do not add that error is approximate and only will the sample numerical value between the precalculated position be made as the numerical value of adjacent sample, and perhaps error is approximate can be by approximate selected and be added to the least significant bit of the 3rd digital data sets from one group of limited error.
In an embodiment of this method, on behalf of first audio signal and this second digital data sets, this first digital data sets represent second audio signal.
Through applying the present invention to audio signal, not only realized can with acceptable accuracy recover first and second audio signals and also realized as the gained combined audio signal by the representative of the 3rd digital data sets be when mixing with second audio signal first audio signal, can be by acceptable expression aspect the perception.Therefore realized correctly not reproduce gained the 3rd digital data sets there being ability to extract from the 3rd digital data sets on the equipment of first or second digital audio and video signals, thereby and had the ability to carry out the equipment that extracts and to extract that first and second audio signals are reproduced discretely or further handle.When using the present invention's combination, when promptly mixing, can also extract only one of them audio signal, and keep another audio signal to be combined more than two audio signals.These remaining audio signals still provide the reproducible audio signal of the mixing of the audio signal that representative still is combined, and can handle the audio signal that has been extracted individually simultaneously.
As a kind of instrument that is used for the sound(-control) engineer-(mix, real-time simulation mix) is possible to the audio mixing of single passage for paired voice-grade channel.During the record editor as a part of verification process, this will form audio frequency output, and this audio frequency output will be represented the minimum minimum quality of ensuring the quality of products and separating audio mixing or decoding channels of final audio mixing process.In case basic AURO-phonic multichannel PCM data set is able to produce, and just can calculated off-line be used to increase the other coding parameter of audio signal quality, thereby need not real-time processing.
In the further embodiment of this method, first seed specimen is first sample of first digital data sets and second sample that second seed specimen is second digital data sets.
Playing the seed specimen that point selection is used to separate near digital data sets allows in case the 3rd digital data sets begins to be read and just begins to separate first and second digital data sets.This seed specimen can also be embedded further, is positioned at this seed specimen sample before thereby promptly be positioned the concentrated recursive scheme that will need of the 3rd numerical data with separation.When the initial number data set begins, perhaps before this, select seed specimen to simplify the separation process that is used to recover first and second digital data sets from this initial number data set.
In the further embodiment of this method, first seed specimen and second seed specimen be embedded into the 3rd digital data sets sample than in the low order.
Through sample than low order in embed seed numerical value; The sample that is affected will only depart from from initial value a little; Have been found that this is actually imperceptible because seed numerical value only seldom need be stored and therefore only seldom sample be affected.In addition, select to have guaranteed only little departing to take place than low order.
Even when the least significant bit of all samples all is used to embed data, this deviation is not discovered yet or is difficult to and discovered, and this is removed from sample because of least significant bit and this is to be difficult to be noted.
Thisly remove least significant bit from sample and reduced and be used to be stored in comprising the required space of the digital data sets of these samples, and therefore on the record carrier or in transmission channel, discharging more spaces or for example allowing to embed other data for the purpose of controlling.
From the PCM sample than low order the other data of encoding read or even during as part of the higher significance bit of the PCM sample that is used for audio frequency; When reading error takes place, use basic skills of the present invention to separate audio mixing PCM sample and can cause error.The essence of this separation process is such, makes will to cause that with these relevant errors of (audio/data) sample the audio mixing of separating of sample subsequently operates.Yet; Be used for using the auxiliary data zone that is used for other data at PCM stream with optimal way; Wherein higher level code will use this auxiliary data zone with storage (sample frequency reduction) error; And make all this correction datas be compressed, with add at place, the end of data block CRC check with so that decoder can be checked the integrality of all data in this.Through storing seed numerical value at regular intervals, can limit because the influence that the error in the audio samples causes.When error takes place, this error will only be transmitted to its known the next position of seed numerical value, and this is because at that some place, can restart separation process, thereby stop error propagation effectively.In addition; In when, in being stored in than the seed numerical value in the auxiliary data zone of low order data error taking place; Separation based on those bad seed numerical value will be wrong; But this only is limited to before the known the next position of its seed numerical value, because at that some place, can restart separation process.
Through sample than the auxiliary data zone in the low order in the other data of storage; The present invention is in the situation of BLU-RayDVD or HD-DVD; The audio mixing of audio mixing voice data (degree of precision position) and encoding/decoding data (each sample is 2,4 or 6 usually) perhaps " multiplexing " except 24 in each sample (available) and do not require any extra record space, and it does not require any extraneous information (for example need not the timestamp of any chapters and sections or stream) from the data on CD " navigation " yet.Likewise, in reading dish control (like embedded software place of execution), need not any change by DVD player.Further, need not to carry out any change in order to use the present invention for the standard of these new media formats or add.And then, the reduction of audio samples bit resolution and in least significant bit storing audio decoding/encoding data during utilizing the equipment do not carry out decoding algorithm or system's (for example HD-DVD or BLU-RayDVD player) normal playback, will make the imperceptible any sound artefact of user (artifact).
In the further embodiment of this method, embed synchronous mode in the position of defining with respect to the position of first seed specimen.
Embed synchronous mode allowing to recover first seed specimen, this be because, when detecting synchronous mode, the position of just known first seed specimen.This also can be applied to locating second seed specimen.Thereby can use flywheel to survey through repeat this synchronous mode at the interval of rule, can further improve this synchronous mode to survey this synchronous mode reliably.This will be divided into piece than the storage in the low order, and this allows to use block-by-block and handles.
In the further embodiment of this method, before the step of sample such as grade, be similar to the caused errors of change such as sample through Select Error from a grouping error is approximate.
Be easy to very much carry out during combination first and second digital data sets Deng the step of changing sample, but also introduced error.
In order to reduce this error, set up error numerical value, from approximate this error numerical value of selecting of one group of limited error that is used for therefrom selecting.
The limited error of this group is similar to and allows to reduce error; And practiced thrift the space simultaneously; This be because, can be only from one group of limited error is approximate, select this error approximate, the limited error of this group is similar to representing still less that can be utilized in etc. that actual error runs into during the change step.Compare with the figure place that during cataloged procedure, discharges, the index that error is approximate requires position still less for each sample.For the compressibility that guarantees data, this is important.This space that is able to practice thrift allows to embed other information for example synchronous mode and seed specimen.The reduction of the sample frequency from 96kHz to 48kHz or from 192kHz to 96kHz possibly become problem; This be because; Introduced higher sampling rate for the purpose that forms audio frequency again; Wherein reproduce, compare with the compact disc audio record for HD Audio, not only such sampling rate and also mainly phase information all be more specifically to need.
To be used for the correction data (error is approximate) that (as much as possible) eliminate these errors can be the result of optimization algorithm owing to sample frequency reduces the sum of errors cause; Wherein, optimization criterion can be defined by the minimum of square error and or even can comprise the criterion based on the sensing audio target.
In the further embodiment of this method; Set up for sample error approximate after; This sample will be waited the numerical value of the adjacent sample change into to be modified, thereby when when comprising the approximate reconstructed sample such as sample such as gradeization of error, this sample is more closely represented at sample before such as grade.As required; Thereby the numerical value through revising adjacent sample when this sample by wait adjacent numerical value when changing into adjacent sample with error approximate combination represent more exactly implementing its adjacent sample etc. change initial sample numerical value before, this error can be by further reduction.
In the further embodiment of this method, this grouping error is approximate to embed the index of representing this error approximate by index and in the approximate corresponding sample of this error.
In the further embodiment of this method, sample is divided into and embeds this index in piece and the sample in first of second front that comprises this index corresponding sample.
Through one group of limited error of index approximate and only the sample of the 3rd digital data sets of this index corresponding sample front than low order in the suitable index of storage, further realized the reduction of the size that error is approximate.Through in previous sample, embedding index, when the separation process of respective sample begins, this index and therefore this error is approximate is available.
In the further embodiment of this method, embedded error is approximate to be compressed.
Except index, can adopt other method that is used to compress, for example LempelZiff.Error is approximate to be similar to and therefore can be compressed from one group of limited error, and this allows use less space when the embedding error is approximate in sample.
If other embed data also be present in sample than in the low order, this is useful especially.Other hereto data, index need not to be available, and can use common compression formats.Can use about the approximate index of error with about the combination of the compression of other data, perhaps can use, be i.e. the overall compression of the approximate and other data of error for all data that in than low order, embed.
In the further embodiment of this method, embed error numerical value with predetermined skew.
Predetermined migration has been set up the relation of confirming in error between the approximate and approximate corresponding sample of this error.
In using the approximate situation of index stores error, this index is applicable to that each piece and suitable index also are stored in each piece.
If possible, index can also be selected or fixing and be stored in the encoder and be not stored in the data flow for each digital data sets, and this is cost with the flexibility.
When not using any error to be similar to the quality of improving the audio signal that is extracted, it is approximate to need not memory error.This do not stop digital data sets than low order in embed and compress other data.
In the further embodiment of this method, embed this error numerical value at the first available position place that has change location with respect to the pairing sample of error numerical value.
Through in case the just error numerical value in the compression samples when having free space; Sample space is able to practice thrift; Can use this space allowing the limited error numerical value of this group of expansion afterwards, thereby permission corective equalization sample has more accurately thus been realized the better reproduction of digital data sets like this.
This can become the method that obtaining space is used, but preferably takes a kind of different scheme.
In fact being used to from the space of compressed error numerical value & index saving to limit will be by the number of samples of audio mixing to together next piece.Because this number is less than current block, littler and therefore various error will can be utilized the error numerical approximation of similar number and be similar to better.The space that these error numerical value and reference key are compressed once more and are practiced thrift continued on for being limited in once more in the next piece by the number of audio mixing sample.
In the further embodiment of this method, be not used to embed any of the approximate perhaps sample other control data, the 3rd digital data sets of error than low order, be set as predetermined value or be set as zero.
Can or before combining digital data sets, perhaps embed for example seed numerical value of embedded information than low order, be set as zero after synchronous mode and the error numerical value.
Predetermined value or remainder value can help to distinguish embedded data, are surrounded because embedded data no longer seems data at random.
It further allows to simplify combination and separation process, because obviously these positions need not to handle.
Be noted that and in than low order, select d/d figure place dynamically to be carried out, in other words, this is based on digital data sets in this instantaneous content.For example the noiseless part of classical music possibly require more position being used for signal resolution, and the loud part of pop music maybe and not require so much position
In one embodiment of the invention; Signal that has been extracted or embedded Control data can be used to control and need the external equipment controlled with audio signal with being synchronized; Perhaps for example through with respect to foundation level or with respect to not by other voice-grade channel that extracts from composite signal, the amplitude of the audio signal that perhaps has been extracted with respect to the combining audio signals definition and control the reproduction of the audio signal that has been extracted.
The invention describes a kind of technology; Be used for usually from 3 dimension audio recordings; But be not limited to this purposes-audio frequency PCM track (the PCM track is a digital data sets of representing digital audio channel)-audio mixing (and storage) is become the track of some, this number is less than the number of tracks of in original records, using.This combination of channels is through to support that reverse operating is that the mode of decode operation becomes single track to realize paired audio track audio mixing; This decode operation allows to separate composite signal; To rebuild the audio track of initial separation; This audio track will with from master tape record (master record; Master recording) original audio track is that perception is identical, and at the same time, composite signal provide a kind of can via regular playback channels reproduce and when by reproduction with the identical audio track of voice-grade channel audio mixing thing perception.Like this; When the channel group of 3 dimension audio recordings (recording) being synthesized one group of passage that is generally used for 2 dimensions around audio recording; And reproduce the passage be combined and when not using reverse operating, be combined, be i.e. (downwards) audio mixing; Audio recording still meet be used for rebuilding be commonly called stereo; 4.0,5.1 or even 7.1 around the requirement of 2 dimensions audio format, true to nature around audio recording, and can be play and not need additional apparatus, modification equipment or decoder by former state ground.This has guaranteed the downward compatibility of gained combining passage.
Expansion to more than 2 digital data sets or two audio signals is very feasible.Released this technology about 2 numerical data collected explanations or commentaries; Can realize in a similar fashion that this technology is to the expansion more than 2 digital data sets; This be through: change staggered; Thereby for each sample of the 3rd digital data sets, only digital data sets provide with quilt with wait the change sample from the sample of the grade combination of other digital data sets non-and select to provide the digital data sets of change sample such as non-from digital data sets that sample is provided with the mode that replaces.
If be combined more than 2 digital data sets; Then every n sample of each digital data sets is used as the sample such as gradeization that the individual data set sample of every n (equating) keeps first subclass of (n-1) individual sample, and the every n of second subclass data set sample keeps 1 sample.For each data set, 1 position of in time domain, squinting, the position of sample such as grade.
Like this, in the data rate and resolution that provide by present DAB standard, have been found that 3 channel digital audios are feasible certainly to 1 channel digital audio audio mixing thing (3 to 1 audio mixing).In this way, 4 to 1 audio mixings also are possible.
This audio mixing of digital audio channel allows to use the first DAB standard of the independent digit voice-grade channel with first number to have the second DAB standard of the independent digit voice-grade channel of second number with storage, transmission and reproduction, and wherein second number of digital audio channel is higher than first number of digital audio channel.
Method of the application of the invention of the present invention or encoder according to the present invention are combined into single digital audio channel with at least two digital audio channels and have realized this point.Because interpolation step in the method, gained digital audio stream are the satisfied expressions of the perception of combined two digital audio channels.
Carry out this combination for a plurality of passages and reduced number of active lanes, for example from 3D 9.1 systems to 2D 5.1 systems.This can be through for example being combined into prepass under the left side of 9.1 systems and upper left prepass usually and can being achieved through left front passage storage, the transmission of 5.1 systems and a left front passage that reproduces.
Therefore, though the signal that uses the present invention to produce allows to recover initial 9.1 passages through separating composite signal, this composite signal is applicable to comparably by the user who only has 5.1 systems and uses.For suitable downward audio mixing 5.1 systems, possibly require two passages of before audio mixing or coding, decaying, thereby during decoding, require (oppositely) attenuation data of each passage.
Use the technology that forms in the present invention-but be not limited to this purposes-to produce the AURO-phonic audio recording that can on HD-DVD that existing or new media bearer for example provides as just instance or BLU-RAYDVD, store; And need not to increase any extra media formats or interpolation to their media formats definition; Because these standards have been supported multi-channel audio PCM data, 6 passages of 24 pcm audios of 8 passages of 24 pcm audios of 6 passages of 24 pcm audios of 96khz (HD-DVD) or 96khz (BLU-RayDVD) or 192khz (BLU-RayDVD) for example.
For the AURO-phonic audio recording, compare with available passage on these existing or new media bearer, need more passage.The present invention allows to use these media bearer that wherein exist passage not enough; Perhaps other transmitting device and make it possible under the not enough situation of the number of active lanes that is used for 3D audio storage or transmission, use this system; And guaranteed simultaneously and all existing playback equipment back compatibles; Thereby automatically in the 2D system, 3D being provided voice-grade channel, is that the 2D voice-grade channel is the same as it.If there is the playback equipment that is suitable for, then can uses according to coding/decoding method of the present invention or complete this group 3D voice-grade channel and this system of decoder extraction and can after extracting the digital audio channel that separates and reproducing these individual passages, complete 3D audio frequency be provided suitably.
Aurophony specified and can correctly provide-by its audio frequency (perhaps audio frequency+video) playback system of recording studio three-dimensional property of x, y and the definition of z axle.Have been found that with the combined suitable SoundRec of (one or more) special loudspeaker layout more natural sound is provided.
The 3D audio recording for example Aurophony can also be defined by have the height loud speaker around setting.The interpolation of this just height loud speaker caused for present normally used system the demand of comparing more passage that can provide, this is because the 2D system that uses at present only is provided at the indoor loud speaker that is in par basically.Because Aurophony merges and fusion the tonality feature in two spaces, so it is associated with specific consciousness aspect.The passage that number increases and the location of loud speaker allow to carry out on this basis the playback of any record with whole potential of the intrinsic three-dimensional aspect of realization use audio frequency.Acoustically take the audience to the actual place of sound event-take the Virtual Space to the combined multichannel technology in special loud speaker location, and making them can in Virtualization Mode, experience its Spatial Dimension.The width in this space, the degree of depth and highly practically and on emotion all by perception first.
And then; Equipment for example HD-DVD or BLU-RayDVD player has been realized a kind of Audio Mixing Recorder; With during playback with external audio passage (not reading) audio mixing from dish to audio frequency output, perhaps usually according to user's navigation operation audio mixing audio frequency effect to increase user's impression.Yet they also have " film " actual pattern, and this pattern has been eliminated these audio frequency effects during playback.Last this pattern by these players be used for through they audio frequency (A/D) transducer output multichannel PCM audio mixing or be used to provide as the audio multichannel audio mixing that in comprising the data of video for example, encapsulates and encrypted multichannel PCM audio mixing and use the HDMI interface to send with further processing.Provide for any equipment or write down and all sets up for the multichannel pcm audio track of these downward audio mixings for the requirement of the lossless compress of during playback/record, using (the for example identical audio frequency PCM data in position), as long as this decoder-as being used to rebuild perhaps " space " enhancing audio recording only of 3 dimension audio recordings in the present invention with explaining.
Except that passing through with reversible manner with the audio frequency PCM storage synthetic single passage of a plurality of channel group, effective or efficient more; One objective is used or used is the audio frequency PCM storage of 3 dimension audio recordings and reproduction, and it still keeps and compatibility like the audio format that provided by DVD, HD-DVD or BLU-RayDVD standard.At control ring during audio recording or multi-channel audio; The record engineer have at present a plurality of audio tracks can with and use template so that their control tool form can for example on CD, SA-CD, DVD, BLU-RayDVD or HD-DVD, create or only be stored in stereo perhaps (2 dimension) on the recording equipment (like, hard drives for example) around audio track with digital form.The audio-source that in real world, always is arranged in 3 dimension spaces so far mainly has been used as the source record that limits at 2 dimension spaces; Even for the audio recording engineer; The 3rd dimension information also is available or can be easily added (for example, aircraft or the bird skyborne " sing " of sound effect as on the audience crown, skimming over) or quilt from real life scene record.
Until now, except following system, the series of other a plurality of audio tracks is stored in the system of track of the abundant number that is provided for storing independently in wherein for example using at the cinema, and it has been available having no the universal audio form.Yet these other passages can not be stored on recording medium such as HD-DVD or the BLU-RayDVD, and this is because these storage systems provide the voice-grade channel of number deficiency.The objective of the invention is to; To can not disturb (perhaps bothering) (2D) mode of standard multichannel or 2 channel audio information with them, with for the record engineer before accomplishing the 3D audio recording basic Real-Time Evaluation be available mode and form these extra " virtual " tracks with the mode of on these new medium, still using no more than " standard " multichannel track.
Should be noted that; Though described the present invention to voice applications; But for Video Applications; Can imagine and adopt identical principle, for example forming 3 dimension rabbits, this is 2 synchronization videos streams (angle) through the camera that uses each all to be taken to have minute angle difference for example; Thereby the formation 3D effect, yet thereby as the present invention describes in detail these two video flowings of combination and therefore make it possible to storage and transmission 3D video it still can playback on the common video equipment.
Application example
Stereo (art) audio mixing that in around audio mixing (SurroundMix), comprises.
During control audio record, sound engineer definition or use the audio mixing template, beginning from a plurality of audio tracks, form " truly " perhaps " art " and stereo-mixing, and around audio mixing (for example 4.0,5.1...).Though is possible around audio mixing to the downward audio mixing of the matrix of stereo-mixing, illustration goes out the shortcoming of this downward audio mixing Matrix Technology at an easy rate.The downward audio mixing of matrix is stereo will to be different from " art " stereo-mixing fully; Because the content from the downward audio mixing stereophonic signal of this matrix will be usually located in the L-R territory (out-of-phase signal); And true " art " stereo-mixing will mainly be arranged in L+R territory (in-phase signal), and in the L-R territory, have the quantity of appropriateness.As just an instance; In single-tone, the downward audio mixing of matrix is stereo will to sound quiet more basically, and this is because the quantity of out-of-phase signal is higher.Therefore, if utilize control of most current audio coding/decoding technology and coding current around audio recording provide usually-they are concerned about true (" art ") stereo version that separates of stereophonics-this record true to nature.
Be utilized in the application of technical foundation of the present invention; Those skilled in the art can easily set up a kind of system; This system is controlled to left and right passage with a left side (preceding) audio frequency of art record with right side (preceding) voice-grade channel, and makes each and (for example) 24dB attenuation audio Delta passage in these passages (L-art-L-around) and (R-art-R-around) audio mixing.When not utilizing any decoder ground to play the L/R passage of multiple recording; To mainly have an art left side/right audio recording, but when the decoder that utilizes as explain is in the present invention play, the audio mixing passage will at first be separated audio mixing; Then; (delta) passage will and be reduced from " art " passage by (for example) 24dB amplification, with for form a left side and right passage as required around audio mixing, also play at this moment around (L/R) passage and center and subwoofer (Subwoofer) passage.
3-dimension (" AURO-phonic ") audio mixing is included in around in the audio mixing.
The coding techniques that uses as explain in the present invention; Can easily see; Simply through 2 the dimension 2.0,4.0,5.1 or even 7.1 around each passage of audio mixing on audio mixing; Another voice-grade channel of the audio frequency of representing as writing down at the certain height place that is higher than those 2 dimension loud speakers, the audio mixing of the 3rd dimension audio-frequency information can be achieved.When not with as this decoder that limits in the present invention when using multiple recording, during audio mixing, these the 3rd dimension voice-grade channels can be attenuated, to avoid nonideal audio frequency effect.During decoding, these passages are separated audio mixing and when needs, are exaggerated, and on the loud speaker of top, these passages are provided.
Stereo (" art ") Hun Yin &3-D (" the AURO-phonic ") audio mixing that in around audio mixing, comprises.
If be intended to produce for artistic stereophonics; The 3-DAURO-phonic reproduction is useful All-in-One record to 2-D around reproducing perhaps; For example 6 passages under 96kHz (HD-DVD) or 192kHz (BLU-RayDVD) then can use based on application of the present invention.Through reducing " initially " sampling rate with the factor 3 (perhaps bigger), the present invention can be used to 3 passages (perhaps more) audio mixing is become a passage, and is similar to the error that during this reduction, produces, to recover initialize signal as much as possible.This can be used to audio mixing 96kHz left front-artistic passage, with the left front Delta of 96kHz (decay) (L-art-L-around) and with the left front top of 96kHz (decay).Similarly the audio mixing scheme can be applied to right front passage.2-passage audio mixing can be applied to a left side around be applied to right around.Even central passage also can be used to top, audio mixing center voice-grade channel.
From " allusion " 2-D record automation 3-D is provided audio frequency.
Audio frequency that great majority exist at present or video production have 2 dimensions (around) audio track.Except that real the 3rd dimension audio source location; As the encoder of explaining in the present invention can use this position during control and audio mixing; Using this information as being become the other passage of 2-dimension record by downward audio mixing, as the diffusion audio frequency that is present in the standard 2 dimension audio recordings is the candidate's audio frequency that moves and provide on the top loud speaker of 3 dimension audio setting.Can expect automation (off-line or non real-time) audio process; This audio process will be extracted the diffusion audio frequency from 2 dimension records; And can use this audio frequency that is extracted forming quilt and 2-D passage, thereby acquisition can be as the 3D audio frequency and decoded around multiple recording around " reductions " audio track audio mixing (according to form of the present invention) of record.According to calculation requirement, can be used in real time around this filtering technique that passage extracts the diffusion audio frequency from 2D-.
The present invention can be used to form several kinds of equipment of the part of 3 dimension audio systems.
Aurophonic encoder-computer application (software) plug-in unit.
Usually the control and the audio mixing instrument that can be used for the V recording and the control world allow the third party to develop software package.They provide common data/command interface to be enabled in by the plug-in unit in audio mixing and the employed kit of Control Engineering teacher usually.Because the core of AUROPHONIC encoder is a simple encoder example; Utilize a plurality of voice-grade channel inputs and a voice-grade channel output on the one hand and be provided with for user on the other hand to take in, can in these audio frequency control/audio mixing instruments, software package be provided like quality and channel attenuation/position as other parameter.
AUROPHONIC decoder-computer application (software) plug-in unit.
Can be to develop software package decoder as the instruments of inspection that utilizes control and audio mixing instrument with the similar mode of encoder plug-in unit.This software package decoder can also be integrated in the media player (like windows media player WindowsMedia Player, perhaps DVD software player and most probable HD-DVD/Blu-Ray software player) of consumer/terminal use PC.
AUROPHONIC decoder-the be built in special-purpose ASIC/DSP in BLU-Ray or the HD-DVD player.
Several kinds of new medium HDs have defined and can flow at their the inner a plurality of high frequencies that use of other (consumer) player of branch/high bit resolution audio frequency PCM.When use wherein have no audio frequency PCM data by audio mixing/fusions/decay/... with the pattern that is provided for the internal audio frequency digital analog converter during from these dish play content, these audio frequency PCM data (can be the AURO coded data) can one group of perhaps for example other top L/R of an art left side/right audio frequency exports for example to transmit to decode the audio mixing voice-grade channel all and to produce one group of extra audio frequency output by special-purpose ASIC or DSP (being loaded with AURO decoder firmware) intercepting.
As one of BLU-Ray or HD-DVD firmware partly integrated AUROPHONIC decoder.
During BLU-Ray or the playback of HD-DVD dish; As long as the AUROPHONIC decode procedure has meaning; The playback mode of these players just need be set as true FiIm pattern, with the Audio Mixing Recorder damage/modification that prevents player as on this dish the primary data of the PCM stream of control.In this pattern, do not need the complete disposal ability of the CPU or the DSP of player.Like this, can be as the other audio mixing process integration AUROPHONIC decoder of separating, carry out this other audio mixing process of separating as a part of the firmware of the CPU of player or DSP.
AUROPHONIC decoder-ASIC/DSP is attached to the HDMI switch, in USB or the FIREWIRE audio frequency apparatus.
HDMI (high definition media interface) makes it possible to transmit the multi-channel audio stream (8 passages, 192kHz, 24) of full bandwidth.The HDMI switch is through the digital audio/video data of regenerating of descrambling at first, thus can inter access in this switch via the voice data of HDMI interface transmission.Can utilize the add-in card of realizing the AURO decoder AURO coded audio of decoding.Similarly additional integrated (usually in audio recording/playback instrument) can be used to USB or FIREWIRE multi-channel audio I/O equipment.
As the encoder of here describing can be integrated in bigger equipment for example in the register system or can be a kind of absolute coding device that is coupled to register system or mixer system.This encoder can also be implemented as computer program for example when moving on the computer system that is being applicable to the said computer program of operation, to carry out coding method of the present invention.
As the decoder of here describing can be integrated in the for example output module in playback apparatus of bigger equipment; In the input module in multiplying arrangement or can be a kind of independent decoder, this decoder be connected (coupling) via its input and flows the source and be coupled to amplifier via its output to the data splitting that has been encoded.
Digital signal processing appts should be understood to be in the equipment in the recording section of record/transmission/reproduction chain in the document, for example the audio mixing form, be used for the recording equipment, signal handling equipment or the signal capture equipment that for example write down on CD or the hard disk at recording medium.
Reproducer should be understood as that it is the equipment in the reproducing part of record/transmission/reproduction chain in this article, and for example audio frequency amplifier or playback apparatus are used for recovering (to fetch, retrieve) data from storage medium.
Reproducer or decoder can advantageously be integrated in vehicle for example in car or the bus.In vehicle, the passenger is surrounded by passenger compartment usually.
This compartment allows easily to locate loud speaker, will be through this loudspeaker reproduction multi-channel audio.Therefore, the designer can ad hoc be customized to audio environment and be suitable at inner 3 dimension or other multi-channel audios that reproduce of passenger compartment.
Another benefit is, as other distribution is hidden, can easily hide the required distribution of loud speaker.The lower set loud speaker of location 3 dimension speaker systems for example is installed in the door-plate as a lot of loud speakers, at present in instrument board or near base plate in the bottom of passenger compartment.Can be for example near roof or be higher than fascia or instrument board or be higher than another position and the upper group loud speaker of this 3 dimension speaker system of location in the top at passenger compartment of lower set loud speaker at least.
Allowing the user also is useful with reproducer from second state that decoder separating audio passage wherein and first state that will separated voice-grade channel be sent to amplifier switch to the voice-grade channel arrival amplifier that wherein has been combined.Can be implemented in the switching between 3 dimension reproductions and the 2 dimension reproductions through walking around decoder.
In the another kind configuration, also be susceptible to the switching between 2 dimension reproductions and stereophonics.
For the requirement of 2 and 3 dimension audio reproducings, the location of loud speaker for example is not a part of the present invention and therefore will be not and detailed description.Yet should be borne in mind that the present invention can be adapted to any channel arrangement that the designer of multi-channel audio reproducer for example can select when car is configured to correctly reproduce multi-channel audio.
Description of drawings
To the present invention be described based on accompanying drawing now.
Fig. 1 illustrate be used to make up two passages, according to encoder of the present invention.
Fig. 2 illustrates by first digital data sets of sample conversion such as gradeization.
Fig. 3 illustrates by second digital data sets of sample conversion such as gradeization.
Fig. 4 illustrates two gained digital data sets is encoded into the 3rd digital data sets.
Fig. 5 illustrates decode the back digital data sets of (one-tenth) two separation of the 3rd digital data sets.
Fig. 6 illustrates improved conversion to first digital data sets.
Fig. 7 illustrates improved conversion to second digital data sets.
Fig. 8 illustrates two gained digital data sets is encoded into the 3rd digital data sets.
Fig. 9 illustrates decode the back digital data sets of two separation of the 3rd digital data sets.
Figure 10 illustrates an instance, the sample of first stream A that has wherein described to obtain like the coding that utilizes as in Fig. 6, describe.
Figure 11 illustrates an instance, the sample of first stream B that has wherein described to obtain like the coding that utilizes as in Fig. 7, describe.
Figure 12 illustrates by the sample of the stream C of audio mixing.
Figure 13 illustrates the error that is incorporated into PCM stream by the present invention.
Figure 14 is illustrated in the form than the zone of the auxiliary data in the low order of the sample of combined type digital data set.
Figure 15 illustrates the more details in this auxiliary data zone.
Figure 16 illustrates and wherein adapts to a kind of situation that causes variable-length AURO data block.
Figure 17 provides the overview like the combination of process steps of explaining in the former part.
Figure 18 illustrates the Aurophonic encoder device.
Figure 19 illustrates the Aurophonic decoder apparatus.
Figure 20 illustrates according to decoder of the present invention.
Embodiment
Fig. 1 illustrate be used to make up two passages, according to encoder of the present invention.Encoder 10 comprises change unit 11b such as the first unit 11a such as grade and second.Each 11a such as unit such as gradeization, 11b all else import receiving digital data collection from the branch of encoder 10.
The first unit 11a such as grade selects the first sample subclass of first digital data sets and each sample of this first subclass etc. is changed into the adjacent sample of the second sample subclass of first digital data sets; Wherein, As will be in Fig. 2 illustrated in detail ground, the first sample subclass and the second sample subclass are staggered.Comprise second subclass unaffected sample and first subclass etc. the gained digital data sets of change sample can be passed to the first optional sample-sized and reduce device 12a or can directly be sent to combiner 13.
The second unit 11b such as grade selects the 3rd sample subclass of second digital data sets and each sample of this three subsetss etc. is changed into the adjacent sample of the 4th sample subclass of second digital data sets; Wherein, As will be in Fig. 3 illustrated in detail ground, the 3rd sample subclass and the 4th sample subclass are staggered.Comprise the 4th subclass sample and three subsetss etc. the gained digital data sets of change sample can be passed to the second optional sample-sized and reduce device 12b or can directly be sent to combiner 13.
First and second sample-sized reduce device and all remove the next of definite (definition) number from their sample of other digital data sets of branch, for example through removing four least significant bit 24 samples are lowered into 20 samples.
As by etc. the changes such as sample carried out of change unit 11a, 11b introduced error.Alternatively, through sample such as grade relatively with initial sample by error approximator 15 approximate these errors.As explaining below, this error is approximate can be used to recover the initial number data set more exactly by decoder.Combiner 13 adds the sample of first digital data sets to the respective sample of second digital data sets; As be provided to its input; And the output via it is fed to formatter 14 with the gained sample of the 3rd digital data sets, formatter 14 the 3rd digital data sets than low order in embed other data for example from the seed numerical value of two digital data sets with as the error that receives from error approximator 15 is approximate and the gained digital data sets is provided to the output of encoder 10.
For interpretation principle, use two inlet flows that embodiment is made an explanation, but the present invention can use three or more inlet flows that are combined into a single output stream comparably.
Fig. 2 illustrates first digital data sets by the sample conversion such as gradeization.First digital data sets 20 comprises sample numerical value A
0, A
1, A
2, A
3, A
4, A
5, A
6, A
7, A
8, A
9Sequence.First digital data sets is divided into the first sample subclass A
1, A
3, A
5, A
7, A
9With the second sample subclass A
0, A
2, A
4, A
6, A
8
Subsequently, as schematically by the arrow among Fig. 2 institute, each sample A of the first sample subclass
1, A
3, A
5, A
7, A
9Each numerical value waited and to be changed into adjacent sample A from second subclass
0, A
2, A
4, A
6, A
8Numerical value.Particularly, this means sample A
1Numerical value by adjacent sample A
0Numerical value substitute i.e. sample A
1Numerical value waited and to be changed into sample A
0Numerical value.This has produced the first sandwich digit data set 21 as shown in the figure, and this first sandwich digit data set comprises sample numerical value A
0", A
1", A
2", A
3", A
4", A
5", A
6", A
7", A
8", A
9", etc., numerical value A wherein
0" equal numerical value A
0And A
1" equal numerical value A
0Deng.In Fig. 6, an embodiment will be shown, wherein because the figure place in the sample reduces A
0" no longer equal A
0
Fig. 3 illustrates by second digital data sets of sample conversion such as gradeization.Second digital data sets 30 comprises sample numerical value B
0, B
1, B
2, B
3, B
4, B
5, B
6, B
7, B
8, B
9Sequence.Second digital data sets is divided into the 3rd sample subclass B
0, B
2, B
4, B
6, B
8With the 4th sample subclass B
1, B
3, B
5, B
7, B
9
Subsequently, as schematically by the arrow among Fig. 3 institute, each sample B of the 3rd sample subclass
0, B
2, B
4, B
6, B
8Each numerical value waited and to be changed into adjacent sample B from the 4th subclass
1, B
3, B
5, B
7, B
9Numerical value.
Particularly, this means sample B
2Numerical value by adjacent sample B
1Numerical value substitute i.e. sample B
2Numerical value waited and to be changed into sample B
1Numerical value.This has produced the second sandwich digit data set, 31, the second sandwich digit data sets as shown in the figure and has comprised sample numerical value B
0", B
1", B
2", B
3", B
4", B
5", B
6", B
7", B
8", B
9", numerical value B wherein
1" equal numerical value B
1And B
2" equal numerical value B
1, etc.In Fig. 7, an embodiment will be shown, wherein because the figure place in the sample reduces B
1" no longer equal B
1
Fig. 4 illustrates two gained digital data sets is encoded into the 3rd digital data sets.
Make up the first sandwich digit data set 21 and the second sandwich digit data set 31 through the addition respective sample now.
For example, the second sample A of the first sandwich digit data set 21
1" with second sample B of the second sandwich digit data set 31
1" addition.The combined sample C that wins
1Be placed in the second place place of the 3rd digital data sets 40 and have numerical value A
1"+B
1".
The 3rd sample A of the first sandwich digit data set 21
2" with the 3rd sample B of the second sandwich digit data set 31
2" addition.The gained second combined sample C
2Be placed in the 3rd position of the 3rd digital data sets 40 and have numerical value A
2"+B
2".
Fig. 5 illustrates decode the back digital data sets of two separation of the 3rd digital data sets.
The 3rd digital data sets 40 is provided for decoder to be separated in two digital data sets 31,32 that comprise in the 3rd digital data sets 40.
The primary importance of the 3rd digital data sets 40 is illustrated as and keeps numerical value A
0", this is the seed numerical value that during decoding, needs.This seed numerical value can be stored in any position, but during explaining for convenience's sake by shown in the primary importance.The second place keeps having numerical value A
0"+B
0" first combined sample.Because decoder is known the seed numerical value A as returning to from primary importance
0", can be through subtracting each other C
0-A
0"=(A
0"+B
0")-A
0"=B
0" and set up the sample numerical value of the second sandwich digit data set.
This recovers sample numerical value B
0" be used to the reconstruct second sandwich digit data set, but also be used to recover the sample of the first sandwich digit data set.Because numerical value A
0" known now, and known its adjacent sample A
1" have identical numerical value, so can calculate the sample of the 2nd sandwich digit data set now:
C
1-A
1″＝(A
1″+B
1″)-A
1″＝B
1″。
This recovers sample numerical value B
1" be used to the reconstruct second sandwich digit data set, but also be used to recover the sample of the first sandwich digit data set.
Because numerical value B
1" known now, and known its adjacent sample B
2" have identical numerical value, so can calculate the sample of the first sandwich digit data set now:
C
2-B
2″＝(A
2″+B
2″)-B
2″＝A
2″。
This recovers sample numerical value A
2" be used to the reconstruct first sandwich digit data set, but also be used to recover the sample of the second sandwich digit data set.
Can be for remaining sample such this point that repeats as shown in fig. 5.
For the approximate first initial number data set 20; The information processing of can using system known relevant signal recovers the first sandwich digit data set; For example for audio signal, can insert or other known signal reconstructing method reconstruct sample that (sample such as grade) lose that is encoded and decodes through interior.As will illustrating afterwards; Can also store relevant through the error that is introduced in mediumization of signal information and use this control information to approach them at the numerical value that waitizations had before, promptly approach the numerical value ground reconstructed sample that they have in initial number data set 21.
Certainly can recover the sandwich digit data set for each and carry out this point will the sample such as grade to revert to and at the approaching as far as possible numerical value of initial value of the sample of initial number data centralization.
To in Fig. 6,7 and 8 the explanation, 2 initial channel for example are reduced to 18 from 24 in every sample in bit resolution below.Then reducing after sample divides rate, sample frequency is reduced for half the (in this example since 2 voice-grade channels, its each all have identical bit resolution and sample frequency) of initial sampled frequency.Other combination also is possible, for example begins from the X position and is reduced to Y position (X/Y=24/22 for example, 24/20; 24/16 etc.. perhaps 20/18,20/16, perhaps 16/15; 16/14...); According to the requirement of HD Audio, should not be in the bit resolution sample is brought down below 14 ... if more passage is by audio mixing, the basic fundamental of then here describing requires sample frequency is become by audio mixing divided by need the number of the passage of a passage.Many more by the passage of audio mixing, then the actual samples frequency of passage (before audio mixing) will be low more.In HD-DVD or BLU-RayDVD, the initial sampled frequency can be up to 96kHz or even (BLU-Ray) up to 192kHz.Since 2 passages, and the two all is reduced to 48kHz when each sample frequency is 96kHz, and this has still kept the sample frequency in the scope of HD Audio.For film/TV audio quality, even 3 passages of audio mixing, and to be reduced to 32kHz also be acceptable (this be as by the employed frequency of NICAM digital broadcasting TV audio frequency).From real 192kHz recording start, provide the mode of 4 passages of audio mixing, sample frequency is reduced to 48kHz.
Fig. 6 illustrates a kind of improved conversion of first digital data sets.Improve in the conversion no longer the representing initial sample than low order but be used to store for example perhaps other control information of information of seed numerical value, synchronous mode, the relevant error that causes through change such as samples of other information of sample at this.
First digital data sets 20 comprises sample number value sequence A
0, A
1, A
2, A
3, A
4, A
5, A
6, A
7, A
8, A
9Each sample A
0, A
1, A
2, A
3, A
4, A
5, A
6, A
1, A
8, A
9Produced the sample A that blocks or round off thereby all block
0', A
1', A
2', A
3', A
4', A
5', A
6, A
7', A
8', A
9'.Organize 60 truncated sample A as explaining among Fig. 2, handling this subsequently
0', A
1', A
2', A
3', A
4', A
5', A
6', A
7', A
8', A
9', wherein obtain considering than low order, perhaps no longer have the information of relevant sample in fact really.This is organized 60 truncated samples and is divided into the first sample subclass A
1', A
3', A
5', A
7', A
9' and the second sample subclass A
0', A
2', A
4', A
6', A
8'.
Subsequently, like the institute of the arrow in Fig. 6 schematically, each sample A of the first sample subclass
1', A
3', A
5', A
7', A
9' each numerical value waited and to be changed into adjacent sample A from second subclass
0', A
2', A
4', A
6', A
8' numerical value.
Particularly, this means sample A
1' numerical value by adjacent sample A
0Numerical value substitute i.e. sample A
1' numerical value waited and to be changed into sample A
0' numerical value.This has produced the as directed first sandwich digit data set 61, and it comprises sample numerical value A
0", A
1", A
2", A
3", A
4", A
5", A
6", A
7", A
8", A
9", etc., numerical value A wherein
0" equal numerical value A
0' and A
1" equal numerical value A
0' etc.
Be noted that promptly sample rounds off because block, in the first sandwich digit data set 61, produce reserve area 62.
Fig. 7 illustrates a kind of improved conversion of second digital data sets.With the mode identical with the mode that is used for first digital data sets; This conversion can be modified; That is, sample no longer represents initial sample than low order but is used to store for example perhaps other control information of information of seed numerical value, synchronous mode, the relevant error that causes through change such as samples of other information.
First digital data sets 30 comprises sample number value sequence B
0, B
1, B
2, B
3, B
4, B
5, B
6, B
7, B
8, B
9Each sample B
0, B
1, B
2, B
3, B
4, B
5, B
6, B
7, B
8, B
9All blocked, thereby produced the sample B of blocking or round off
0', B
1', B
2', B
3', B
4', B
5', B
6', B
7', B
8', B
9'.
Organize 70 truncated sample B as explaining among Fig. 3, handling this subsequently
0', B
1', B
2', B
3', B
4', B
5', B
6', B
7', B
8', B
9', wherein obtain considering than low order, perhaps no longer have the information of relevant sample in fact really.
This organizes 70 truncated sample B
0', B
1', B
2', B
3', B
4', B
5', B
6', B
7', B
8', B
9' be divided into the 3rd sample subclass B
0', B
2', B
4', B
6', B
8' and the 4th sample subclass B
1', B
3', B
5', B
7', B
9'.
Subsequently, as schematically by the arrow among Fig. 3 institute, each sample B of the 3rd sample subclass
0', B
2', B
4', B
6', B
8' each numerical value waited and to be changed into adjacent sample B from the 4th subclass
1', B
3', B
5', B
7', B
9' numerical value.Particularly, this means sample B
2' numerical value by adjacent sample B
1' numerical value substitute i.e. sample B
2' numerical value waited and to be changed into sample B
1' numerical value.
This has produced the as directed second sandwich digit data set 71, and it comprises sample numerical value B
0", B
1", B
2", B
3", B
4", B
5", B
6", B
7", B
8", B
9", numerical value B wherein
2" equal numerical value B
1' and B
1" equal numerical value B
1', etc.Be noted that promptly sample rounds off because block, in the second sandwich digit data set 71, produce reserve area 72.
Being reduced on the principle through the resolution that is introduced into like rounding off of explanation in Fig. 6 and 7 is " expendable ", but can use the technology that increases the perception sample frequency.If require more bit resolution, then the present invention allows to increase the numerical value (position that in fact is used) of Y, its cost be used for coded data or every sample the X position available " space " still less.The approximate perception loss that allows significantly to reduce resolution of the error of certainly, storing in the data block in the auxiliary data zone.
For 24 pcm audio streams; Utilize 2 passages of 18/6 form and audio mixing; We obtain 18 audio samples and 6 bit data samples; Each data block is all with the beginning of the synchronizing signal (sync) of 6 data samples (each 6), and length and final 2 * 3 data samples (2 * 18) that 2 data samples (12 altogether) are used to store data block are used to the stored copies audio samples.About other form (instance):
The synchronizing signal of-16/8:8 data sample, 2 data samples (16 are only used 12) are used for length, and 2 * 2 data samples (2 * 16) are used for the copy audio samples;
The synchronizing signal of-20/4:4 data sample, 3 data samples (12 altogether, bit) are used for length, and 2 * 5 data samples (2 * 20) are used for the copy audio samples
The synchronizing signal of-22/2:2 data sample, 6 data samples (12 positions altogether) are used for length, and 2 * 11 data samples (2 * 22) are used for the copy audio samples.
About other form (for example 16 pcm audios have 14/2 form), can define similar structure.
Fig. 8 illustrates two gained digital data sets is encoded into the 3rd digital data sets.
With with carry out coding in mode identical described in Fig. 4.
Since the first sandwich digit data set 61 has reserve area 62 and the second sandwich digit data set 71 also has reserve area 72, now two digital data sets additions have been produced the 3rd digital data sets 80 with auxiliary data zone 81.
In this auxiliary data zone 81, can place other data.
When utilizing when not knowing the equipment that has this auxiliary data zone 81 and reproducing the 3rd digital data sets 80, the data in this auxiliary data zone 81 will by this equipment be construed to be with the digital data sets that is reproduced than low order.
Therefore the data that are placed in this auxiliary data zone 81 will be introduced in imperceptible, slight to a great extent noise to signal.This being not aware of property depends on certainly and is selected as the number than low order that this auxiliary data zone 81 keeps, thereby and those skilled in the art's right quantity than low order that will be used of being easy to select be equilibrated at call data storage and the mass loss of concentrating in numerical data that is caused in the auxiliary data zone 81.Obviously, in 24 audio systems, the number than low order that is exclusively used in auxiliary data zone 81 can be higher than the number in 16 audio systems.
In order to make it possible to that these audio mixing voice-grade channels are carried out contrary (perhaps separating audio mixing) operation, the copy of a limited number of sample is stored.
Though in above instance, the copy that only single seed numerical value sample is a sample is used and stores, it is favourable storing a plurality of seed numerical value samples, because redundancy is provided like this.This redundancy is because the repetitive nature of the stored seed numerical value of institute, and this repetitive nature allows through NEW BEGINNING point is provided in stream from the error recovery, and the while promptly can be stored two seed numerical value that are used for each starting position due to the fact that.Seed numerical value A0 and B1 allow the check starting position, and this is that numerical value B0 can be compared to test with stored seed numerical value then because the calculating that begins with A0 will produce numerical value B0.Further advantage is; The two storage of A0 and B1 allows two seed numerical value of search to belong to its correct starting position; Permission between seed numerical value and digital data sets C automatically synchronously, this be because be wherein to use the position of seed numerical value A0 decoding to produce exactly probably to equal the numerical value B1 of stored seed numerical value B1.
When beginning, for example, the 96kHz sampled signals are reduced to 18 (Y) position 48kHz from 24 (Z) position, and every millisecond (msec) produce the copy of a sample, i.e. every millisecond of seed numerical value, each passage audio mixing 1000 18 sample copies, i.e. seed numerical value.If this audio mixing comprises 2 passages, then per second is used for needs 2 * 1000 * 18 or 36K position of sample copy " storage ".Because produced first extra " space "-under per second 96K; Each sample 6 (X) position; So in by the auxiliary data zone that forms than low order, per second 6 * 96=576K position is available, these copy of storing sample numerical value easily wherein.Therefore in fact, existence can be used to store the 16x memory of these copies, and if in this auxiliary data zone, will not store any out of Memory, then can be with every millisecond of copy sample of these 2 passages of speed storage of 16 times.If selected to be used for other numerical value of Z/Y/X, under 96kHz 24/20/4 or under 44.1kHz 16/14/2, will be different then for example through the regional quantity of " freedom " auxiliary data of using least significant bit to produce.Provide following situation with example forms, but the invention is not restricted to these other use situation; At 2 passages of 24/20/4 under 96kHz, and per second 4 * 96=392K bit memory requires every millisecond of 2 * 1000 * 20=40K position to be used for the copy sample, can be with every millisecond of speed stored copies sample of 9.6 times.At 2 passages of 16/14/2 under 44.1kHz, and per second 2 * 44.1=88.2K bit memory requires every millisecond of 2 * 1000 * 14=28K position to be used for the copy sample, can be with every millisecond of speed stored copies sample of 3.15 times.The instance of here addressing will ad hoc be used for reproducing sample from initial (resolution and frequency reduce) audio stream by the auxiliary data zone that forms than low order of sample.Because as the character and the characteristic of the technology of here using, it is useful not only using this " freedom " auxiliary data zone to be used for the stored copies sample, though these sample copies are by separating the essential information that audio mixing process or decoder use.
In basic fundamental,, at first reduce by 2 pcm audio stream A (A as in Fig. 2-8, explaining ground
0, A
1, A
2) and B (B
0, B
1, B
2) bit resolution, with produce 2 new stream A ' (A '
0, A '
1, A '
2) and B ' (B '
0, B '
1, B '
2).Then, the sample frequency of these streams is reduced for the half the of initial sampled frequency, thereby provides A " (A "
0, A "
1, A "
2) and B " (B "
0, B "
1, B "
2).Error has been introduced in this last operation, wherein A "
21=A "
2i+1=A '
2iProduced error E
2i+1=A '
2i+1-A '
2iAnd B "
2i+1=B "
2i+2=B '
2i+1(B "
0=B '
0) produced error E
2i+2=B '
2i+2-B '
2i+1(E
0=0).This error series (E
0, E
1, E
2, E
3...) contain owing to the sampling of audio stream B reduces the sum of errors with even number index that produces because the error with odd number index that the sampling of audio stream A reduction produces.Higher level code will be similar to these errors and use these approximate before audio mixing, to reduce error.As a part of audio mixing, as sample than the auxiliary data zone in the low order in the separate channels set up add approximate error (being expressed as the inverse of real error) E '.Like this, utilize sample (Z=A "+B "+E '), audio signal is defined by Z
i=A
i"+B
i"+E
i'.If error stream can be approximate by exactly, then E '=E, wherein Z
2i=A
2i"+B "
2i+ E
2i=A '
2i+ B '
2i-1+ B '
2i-B '
2i-1=A '
2i+ B '
2i, and Z
2i+1=A
2i+1"+B "
2i+1+ E
2i+1=A '
2i+ B '
2i+1+ A '
2i+1-A '
2i=A '
2i+1+ B '
2i+1In this case, in final audio mixing stream, do not produce any sampling and reduce error.
Fig. 9 illustrates decode the back digital data sets of two separation of the 3rd digital data sets.
Such as the decoding of the routine in Fig. 5, described, carry out through enhance encoding, promptly be used to store other data and the decoding of the digital data sets 80 that obtains, but decoder only provides a relevant A of each sample than low order 81
0", A
1", A
2", A
3", A
4", A
5", A
6", A
7", A
8", A
9", B
0", B
1", B
2", B
3", B
4", B
5", B
6", B
7", B
8", B
9", promptly be not than low order.Decoder can further recover the other data of storage in than the zone of the auxiliary data in the low order 81.As in Figure 20, explaining ground, these other data can be sent to the target of other data subsequently.
In case decoder is with these copy samples, the reconstruct of seed numerical value, then these copy samples (seed numerical value) just are used to separate audio mixing audio mixing passage then.The audio mixing passage for example is the audio mixing of PCM stream A " and B ", wherein A "
2i=A "
2i+1=A '
2iAnd B "
2i+1=B "
2i+2=B '
2i+1A '
0And B '
1To be used as the copy sample and be encoded into data block.
As substituting of the method for in Fig. 5, explaining, wherein only use a seed numerical value, can realize as follows that separating audio mixing (single-tone) signal: A "+B " sample from A "+B " is: A "
0+ B "
0, A "
1+ B "
1, A "
2+ B "
2, A "
3+ B "
3, A "
4+ B "
4, A "
5+ B "
5Because we have A "
0=A '
0&B "
1=B '
1Copy, so we can reconstruct A " &B " stream.
1. utilize A "
0+ B "
0-(A "
0=A '
0), we obtain B from the copy sample "
0And obtain A "
0
2. utilize A "
1+ B "
1-(B "
1=B '
1), we obtain A from the copy sample "
1And obtain B "
1
3. utilize A "
2+ B "
2-(B "
2=B "
1), we obtain A "
2And B "
2=B "
1
4. utilize A "
3+ B "
3-(A "
3=A "
2), we obtain B "
3And A "
3=A "
2
5. utilize A "
4+ B "
4-(B "
4=B "
3), we obtain A "
4And B "
4=B "
3
6. utilize A "
5+ B "
5-(A "
5=A "
4), we obtain B "
5And A "
5=A "
4
7....
On media formats such as HD-DVD or BLU-RayDVD, can be as multiplexing pcm audio stream storage multi-channel audio.On each of these passages, use as, can easily reproduce these a plurality of passages (from 6 or 8 to 12 or 16) at the audio mixing of above explanation/separate the audio mixing technology.This allows to store or form the audio recording or the reproduction of the 3rd dimension through above each floor speakers, increasing the top loud speaker; And not requiring that the user has " 2 dimension " version that decoder is listened to this audio frequency, this is " can play " audio frequency because the audio frequency of on the multi-channel audio track, storing remains 100% PCM.In this last a kind of reproduction mode, with the effect that does not produce the 3rd dimension, but it can not reduce the discernable quality of 2 dimension audio recordings yet.
Figure 10 illustrates an instance, has wherein described the sample like first stream A that obtains through the coding of in Fig. 6, describing.
For example, suppose and to handle 2 single-tone 96kHz24 bit digital audio stream A&B.
The initial sample of A=(24), the A '=quilt sample (18H You Xiaowei &6L position=0) that rounds off, "=sample frequency reduces sample to A.
In Figure 10, in chart, the first audio stream A is shown with the dull gray line.The sample of A is: A
0, A
1, A
2, A
3, A
4, A
5... the resolution of each sample is every sample 24 (Z) position that is expressed as 24 signed integer numerical value, so number range is from-2
(Z-1)To (2
(Z-1)-1).From this series of samples, we reduce to 18 (Y) position with resolution, remove 6 (*) least significant bit to be formed for " space " of coded data.Reduction is through using the only Y highest significant position in Z altogether that the immediate expression that all Z position samples are rounded to them is realized.So far, each sample is coupled with (2
(X-1)-1), each sum is restricted to (2
(Z-1)-1) perhaps is expressed as [] (2
(Z-1)-1).Then, utilize AND ((2 by turn
(Y)-1) the X position that is shifted by turn left), we are made as 0 with the individual least significant bit of 6 (X), and like this, we have produced a new stream A ' (light gray).The sample of A ' is: A '
0, A '
1, A '
2... A ' wherein
i=[A
i+ (2
(X-1)-1)] (2
(Z-1)-1) AND ((2
(Y)-1)＜＜X).
After reducing sample resolution, we also with factor 2 reduce sample frequency (we with the situation of audio mixing more than 2 passages in, we need be to equal by the factor of the number of active lanes of audio mixing reduction sample frequency).So far, we repeat each even samples of initial flow A '.After sample frequency reduced, we obtained new stream A ".A " sample be: A "
0, A "
1, A "
2... A wherein "
2i=A '
2i+1=A '
2i
A at index (subscript) 2i place " all even samples identical with primary data at the A ' at index 2i place, and at the copy of the previous sample of the A at index 2i+1 place " all odd samples be A " at index 2i place.
Figure 11 illustrates an instance, has wherein described as flowing the sample of B through first that obtains like the coding of in Fig. 7, describing.
The initial sample of B=(24), the B '=quilt sample (18H You Xiaowei &6L position=0) that rounds off, "=sample frequency reduces sample to B.
In Figure 11, in chart, the second audio stream B is shown with the dull gray line.Same sample resolution reduction is applied to this stream.The sample of B is: B
0, B
1, B
2, B
3, B
4, B
5... from this series of samples, we produce new stream B ' (light gray).The sample of B ' is: B '
0, B '
1, B '
2... wherein:
B′
i＝[B
i+(2
(X-1)-1)](2
(Z-1)-1)AND((2
(Y)-1)＜＜X)。
After reducing sample resolution, we also reduce sample frequency with factor 2 similarly and we obtain new stream B ".B " sample be: B "
0, B "
1, B "
2... wherein: B "
2i+1=B "
2i+2=B '
2i+1
B at index 2i+1 place " all odd samples all identical with primary data at the B ' at index 2i+1 place, and at the copy of the previous sample of the B at index 2i+2 place " all even samples all be B " at index 2i+1 place.
Figure 12 illustrates the sample of audio mixing stream C.
The initial sample of A+B=(24), the A '+B '=quilt sample (18H You Xiaowei &6L position=0) that rounds off, A "+B "=sample frequency reduces sample.
Two stream A+B by audio mixing (by addition) to obtain new stream (dull gray).Then we obtain another stream (light gray) will to flow A " and B " audio mixing (addition).For each sample, A "+B " will be different from A+B and be different from A '+B ', this be because; Because bit resolution reduces (rounding off); A " or B " can be different from initial sample A and B, and reduces sample owing to sample reduces and can be different from resolution, but usually; Because initial high bit resolution and high sample frequency, the excellent perception that we still have initial A+B (dull gray) stream is approximate.
Figure 13 illustrates the error of being introduced PCM stream by the present invention.
The sample owing to round off, and Error=Errors, because the sample that rounds off+frequency reduction, and Error '=Errors.
Figure 14 illustrates the form than the zone of the auxiliary data in the low order of the sample of combined type digital data set.
Finally, in order to make decoder can separate audio mixing audio mixing audio frequency PCM data, decoder requires before it receives audio frequency PCM sample, to have the copy sample of audio frequency PCM sample, separates the audio mixing operation thereby can utilize streaming audio PCM to carry out in real time.So far, we need place the sample (Z position) that also has the audio frequency PCM information relevant with previous data block with these data (the copy sample, synchronous signal mode, the length parameter that keep audio samples ...) of data block.In order to give the time of these data blocks of decoder decode, they in addition can be used to take to finish several audio frequency PCM samples before the audio frequency PCM sample of copy from it.Number terminal and that be used to as the audio frequency PCM sample between the audio frequency PCM sample of copy sample copy in data block is Offset (skew), and this is another parameter of in data block, storing.Sometimes this skew can be born, and the position of the copy sample of this explanation in audio frequency PCM stream is arranged in the audio frequency PCM sample that is used to carry that data block.For skew, we also will use 12 bit value (signed integer numerical value).
Data block comprises:
1. synchronous signal mode
2. data block length
3. with reference to the audio frequency PCM sample skew at the terminal of that data block.
4. the copy of audio frequency PCM sample (having) for each audio mixing passage.
Negate the control information of the error through change introducings such as samples and realized further advantage through comprising permission (part).
In Figure 14, in the time 0, encoder begins to read 2xU X position sample, and they are reduced to Y position (bit) to be formed for keeping the auxiliary data zone of data block.The sample frequency reduction has produced error, and this error quilt is approximate and quilt is alternative with reference to these approximate tabulations.Except these data that are effectively incompressible--the generation data block head (synchronizing signal, length, skew ... etc.), thereby the data block length of the individual sample of formation U '.These data samples are placed in the data segment of a U sample.In the step below, encoder is read U ', and (＜U) individual sample requires U sample and the " data block of individual sample that after compressing, requires U thereby produce (not being compressed ground).Equally, this data block be coupled to previous data block and in this example (still) use some samples in initial U (X position) sample.Encoder is read U '
..' X position sample and the process continuation that produces corresponding data block all are processed until all data.
Figure 15 illustrates the more details in auxiliary data zone.
AUROPHONIC Data Carrier Format (data carrier format) meets following structure.
It is an accurately audio/data stream 150 of position, is generally PCM stream 150, and wherein data are divided into the section 158,159 of Z sample.Each sample standard deviation in section 158,159 is made up of X position.(for audio frequency CD/DVD data, X will be 16 usually, perhaps for the BIu-Ray/HDDVD voice data, be 24).(Y first of highest significant position; For for example Blu-Ray, 18 or 20 positions normally) keep voice data (can be the pcm audio data), least significant bit (Q last position; For example for Blu-Ray, normally 6 or 4) keep the AURO decoded data.
The other data of AURO that are organized in each data block 156,157 as follows as during decoding, use:
It comprises synchronizing signal section 151, general decoding data segment 154, index 152 and error form 153 alternatively, and is CRC numerical value 155 at last.
Synchronizing signal part 151 is become scroll bits pattern (size depends on the Q bits number that is used for the AURO data width) by predefine.Conventional data 154 comprises the information of the length of relevant AURO data block; Must use the accurate skew (with respect to synchronizing signal position 151) of first audio frequency (PCM) data 158 of AURO decoded data 156 above that; The copy of first audio frequency (PCM) data sample (having) for each channel coding, attenuation data and other data.(select) the error form 153 that all errors that this AURO decoded data 156,157 produces during also can comprising index 152 and remaining on coding step are approximate alternatively according to the AURO quality during cataloged procedure.In addition, also alternatively, index 152 can be compressed with error form 153.Whether general decoding data segment 154 will indicate this index 152 to exist with error form 153, comprise relevant applied information compressed.At last, CRC numerical value 155 is to use the CRC of audio frequency PCM data (Y position) and the two calculating of AURO data (Q position).
A utmost point low latency that is characterised in that it of AURO decoder.Decoding only needs the processing delay of 2 AURO (PCM) sample.Must, transmission AURO decoded data transmit and handle (for example decompressing) AURO data block 156,157 information before need being applied to pcm audio data 158 wherein.As a result, AURO data block 156,157 (least significant bit) quilt always is not later than the AURO data message with the feasible last AURO data message 154,155 from a piece of audio frequency PCM data 159 (highest significant position) fusion and is applied to first (PCM) audio data sample wherein.
The decoder of separating the audio mixing operation of carrying out passage uses synchronous signal mode for example to locate the copy sample and they are associated with the matching initial sample to allow it.For each sample, these synchronous signal modes also can be placed in 6 (X) position, and should detect easily by decoded device." sync (synchronizing signal) " pattern can be the repeat pattern of the sequence of several 6 (X) positions long " keys (key) ".For example, be displaced to highest significant position and put through making single position put from least significant bit, perhaps binary system is expressed as: 000001,000010,000100,001000,010000,100000.But thereby can avoid this synchronous signal mode to influence sample based on other bit pattern of feature selecting of sample with perceptive mode, perhaps this sample influenced synchronous signal mode is surveyed.Like this, can make up for all different sample resolution (24/22/2,24/20/4/, 24/18/6,24/16/8,16/14/2...) the consistent synchronous signal mode of definition.These patterns can also be optimized to eliminate when not used the DVD-player plays of this AURO-Phonic decoder " noise " that produces from the least significant bit of audio samples.
Figure 16 illustrates a kind of situation, the wherein adaptive variable-length AURO data block that causes.Further require decoder before its handles audio mixing audio samples, to receive the information of data block, this be because, it must decoded data block (comprising decompressions) and needs visits these (being similar to) thus the error execution separate audio mixing and operate.With utilization comprise approximate form and reference listing to the error stream sample (from that the 2nd
nPiece) approximate (using K-intermediate value or Facility Location facility location algorithm) is associated with an element of that approximate form with each sample with that error stream section.This reference listing constitutes approximate error stream.This tabulation and the form with numerical approximation all are compressed the device compression, other surplus element of data structure by the formatter definition (like synchronous signal mode, data block length, skew, copy audio samples, decay etc.) (thereby very likely) will finish with the data sample that is less than U, we are called W (W＜=U) with this number of samples.Can expected value W littler by 20% to 50% than U usually.Then, this data block is placed the data space of a U sample by formatter.This has guaranteed that decoder can use these data samples before it receives the coupling audio samples.Because we can practice thrift (U-W) individual data sample so that later use, thus the next audio section of will be encoded (this is that audio mixing and error are approximate) should only comprise W audio samples (＜=U).Should require U data sample even if be used for the data block of this section (W audio samples), also guarantee this data block that before first audio samples of its reference, terminates.And then, because more the peanut audio samples (W＜=U), so we can expect that reducing the approximate of error for sample frequency is better, this is because must the approximate more error numerical value of peanut.Like this, use compression gains by better being similar to of next section of audio samples.Again, the last section of this of data block can be less than U, for example W ' (thereby＜=U) next number with the audio samples that is encoded can and then also be restricted to W '.
Should be appreciated that further according to compression quality, the size of data block will change.Therefore, offset parameter (part of block data structure) is the important parameter that is used for the data block associated of change in size is arrived corresponding first audio samples.First audio samples that is associated with data block from utilizing offset parameter begins, the number of the audio samples that the length coupling of data block self needs during decoding.When decoder in particular condition instantaneously possibly need the more time to begin data block when decoding with respect to what it received the first coupling audio samples, if desired what for to adding this offset parameter (and data block displacement more backward in time).Should be appreciated that further decoder should be carried out the data block decoding at least in real time, this is because this delay can not increase.
Of the present invention another be characterised in that, decoder will easily rest in the synchronizing signal that has the synchronizing signal reference and and then automatically survey the coded format (detection is used to the figure place of the audio samples of synchronous signal mode/sample copy) that has been used.So far, we comprise number of samples as a part of coded data between each first word of synchronous signal mode.We also require synchronous signal mode repeating after the individual sample of 4096 * 2 (2=audio mixing number of active lanes) at the most.This maximum length with data block (synchronous signal mode+sample copy data) is reduced to 4096 * 2 samples, thereby requires 12 positions to store this length of each data block.Use this information, and given different coding resolution is for example for 24 PCM samples: 22/2,20/4,18/6,16/8, decoder should be able to be easily recognition coding form automatically, survey synchronous signal mode and repetition thereof.
Embedding auxiliary data in the data area that forms than low order by sample can use with combination/separating mechanism independently.Equally in single audio frequency stream, can produce this data areas and be not embedded into signal wherein acoustically influencing auxiliary data.If do not carry out any combination; To embedding that error is approximate to remain useful because sample frequency reduces error that changes such as () samples cause; This is because it also allows to reduce sample frequency (so conserve storage), still allows the influence that the approximate reconstruct initialize signal well of use error reduces with the reply sample frequency as explaining.
Figure 17 illustrates and comprises the improved coding of all embodiment.
Shown frame is corresponding to method step and comparably corresponding to the hardware block of encoder and be illustrated in data flow between the hardware block and between method step.
The encoding process step.
In first step, at first through (rounding) audio samples (24 → 18/6) that rounds off with audio stream A, B reduces to A ', B '.
In second step, application of dynamic compression on these streams and stream that premix sound (use attenuation data) has been lowered with avoid the audio frequency amplitude limit (A '
c, B '
c).
In third step, with equal the audio mixing passage (A '
C', B '
C') thereby the factor of number reduces sample frequency introduces error stream E.In the 4th step, utilize E ' approximate error stream E: use 2
(Z-1)Individual center (for example K-intermediate value approximate) and be used for the reference listing at these centers.
In the 5th step, form and reference are compressed, and having defined sampling decay (audio samples begins), and build (synchronizing signal, length ..., crc).In the 6th step, stream (A '
C', B '
C', E ') by audio mixing, comprise that finally (audio frequency overshoot) checks that this inspection possibly require minor variations for amplitude limit.In the 7th step, data block section (6 samples) merges with audio samples.
Figure 17 provides the summary signal like the combination of process steps of explaining in the part in front.Should be appreciated that when in the off-line situation, using, this cataloged procedure is worked the most easily, encoder can be visited the sample of the corresponding part of all stream that it must at any time handle.So, can be searched (ground, front and back) data to use its to need in order to handle that part thereby the part that requires audio stream for example temporarily is stored on the hard disk encoder process at least.In the explanation of Figure 17; Use a kind of like this situation as an example; Wherein 24 samples (X/Y/Z)=(24/18/6) is divided into 18 sample numerical value and 6 bit data numerical value, the part in the auxiliary data zone that this 6 bit data numerical value is retentive control data and seed numerical value.
Block length-, will be known as U for vague generalization.
The first step of cataloged procedure < 1>(like what in the part about basic fundamental, explain) is through each sample being rounded to its immediate 18 bit representations, with flowing on the B161b sample resolution for example being reduced to 18 from 24 at two stream A161a and utilize sample-sized to reduce device.These stream 163a, 163b as this result who rounds off are known as stream A ' 163a and stream B ' 163b.Side by side, utilization is confirmed decay from the attenuator controller that input receives desired decay numerical value 161c.
Second step < 2>is to utilize the decay executor to flow 163a at these, and the last audio mixing that carries out of 163b simulates to analyze audio mixing whether to cause amplitude limit (clip).Before audio mixing, decay one if desired and flow 163b, in the situation of AURO-PHONIC coding, normally the 3rd tie up audio stream, the executor of then decaying should take in this decay in this audio mixing simulation.Although if this decay; Audio mixing two (96kHz) stream 163a, 163b will produce amplitude limit, and this step of the cataloged procedure of then being carried out by the decay executor will be carried out smooth compression (little by little increase the decay of audio samples and follow little by little with its reduction towards clipping point).This compression can be attenuated executor be applied to flow 163a, 163b the two, but this is dispensable, also can eliminate this amplitude limit because (more many) on a stream 163b compress.When being applied to these stream A ' 163a with stream B ' 163b, attenuation controller produces new stream A '
c165a and stream B '
c165b.The effect that is used for preventing this decay of amplitude limit will retain in final audio mixing stream 169, and separate in the audio mixing stream.In other words, decoder will can not compensate this decay with generation initial flow A ' 163a or initial flow B ' 163b, but its target will be to produce A '
c165a and B '
c165b.During control this (Aurophonic) record; If desired, the record engineer can define Reduction Level 161c and via input it offered attenuation controller to control the decay of the second stream 163b (the common the 3rd ties up audio stream) that when being become 2 dimension audio reproducings by downward audio mixing, expects.
Step below<3>In, frequency reduce device with equal the passage audio mixing (A '
C', B '
C') the factor of number reduce sample frequency, thereby introduce error stream E167.Can be for example as in Fig. 2 and 3, that kind of perhaps explaining in 6 and 7 is carried out the frequency reduction.
Step below<4>In, utilize the E ' 162 approximate errors stream E167 that produces by the error approximator: use 2
(Z-1) individual center (for example K-intermediate value approximate) and be used for the reference listing at these centers.
In higher level code/decoded portion, explained to be similar under the condition of this error stream 167 that the error 167 (because sample frequency reduces) of conciliating in the audio mixing operation at audio mixing can be able to avoid error freely.At this instantiation (X/Y/Z)=(24/18/6) and V=32 (2
(Z-1)) approximate in, very possible is, when we only have V sample in data block, does not have any error (restriction that causes except 12 bit representations owing to error), thereby exists these errors to arrive the mapping one by one that these " are similar to ".On the other hand, we have also defined the maximum length U of data block, and this will guarantee that under any circumstance error reference listing and approximate form will " can be encoded " in this data block.Therefore this coding step will be incipiently from two stream A '
c' 165a and B '
c' 165b and be the sample of U from error stream E167 number required.
At first, the width of Select Error sample (this is the figure place that is used to represent this control information).Because basic stream is the PCM data from audio recording, be less so can be expected at 2 error or differences between the adjacent sample and maximum (perhaps minimum) sample is compared.For (for example) 96kHz audio signal, only when audio stream contained the signal with very high frequency, this error can be bigger.As before explain, in this explanation, use 24 PCM to flow, be reduced to the space that 18 positions being used for audio frequency and generation are used for 6 data bit for each sample.As explaining in the basic fundamental; Use these data bit with the length of storage synchronous signal mode, data block, skew, with the parameter that is determined, 2 copy samples (when 2 passages of audio mixing), compressed " error index ", compressed error form and verification and.To explain " error index " and error form below.In 24/18/6 instance, 6 positions of every sample can be used for auxiliary data zone and 6 positions of every sample and can define in theory as required and have 2
6The form of=64 errors.In this instance of 24/18/6, error representes to be restricted to signed 2x6 position integer.
The partial content (there is an audio frequency (audio mixing) sample in 24/18/6-for each sample of data block) that has the data block in 6 the auxiliary data zone of U sample is to have the form that error that the sample frequency owing to these streams reduces to produce is similar to.As before address, will use 26 bit data sample approximate errors.Because there be not enough " space " to be used for the approximate of each error, so need definition as much as possible near the limited numeral of the error ' numerical value of these all errors with storage.Then, produce tabulation, this tabulation comprise error ' stream in the data block that is used for the auxiliary data zone ' the quoting for these approximate errors of each element.Remove synchronizing signal, length, skew, sample copy etc.. outside, need the space with storage in data block have approximate error ' form.This form can be compressed, and is used for the memory of data block with restriction, and and then reference listing also can be compressed.
At first, with probing into from the mode of approximate these elements of error stream.What need be defined is that number is the numerical value of K, thus each element (but normally a part of pairing this stream of data in data block) of stream thus can be as far as possible little by and sum of the deviations related (this is each element of error stream and the antipode of its best (the most approaching) numerical approximation error) with one of these numerical value.Can use other " weighting " factor to replace absolute value, for example this absolute value square or the definition that takes in for the sensing audio characteristic of definition.Owing to reducing K the such numeral of error discovery that produces, the sample frequency of 2 audio mixing passages is defined by K-intermediate value target from series of values-be defined by in this case.Group from the element of error stream needs assembled, and need K center of identification make from each put it nearest center apart from the sum minimum.
For example in the facility location algorithm, similar problem and scheme thereof also are known at document.And then, need to consider " fluidisation " scheme and non-fluidisation scheme in this article.The former will mean that " encoder " only has once for the life-span that produces from life-span audio stream audio mixing (with in real time) error and one visit all over and ask.Latter's (non-fluidisation) will mean that encoder has " off-line " and connected reference for the data that it requires to handle.Because the structure of output digital data stream (audio frequency PCM stream) with 18 audio samples and 6 bit data; Data block from the auxiliary data zone was issued before its pairing audio samples, produced to be used for the state that non-fluidisation is used K-intermediate value or facility location algorithm situation.Because these a lot of algorithms all are available in open source literature, so the object of the invention is not a kind of new packet algorithm of definition, but the scheme that conduct is used for the technology implementation personnel is with reference to these algorithms.[for example see Clustering DataStreams:Theory and Practice (packet data streams: theory and practice), IEEETRANSACTIONS ON KNOWLEDGE AND DATA ENGINEERING, VOL.15, NO.3, in May, 2003/June].
Be determined in case this K center or error are approximate, just produce tabulation, wherein from L element of the error stream of audio mixing by for L of the element in that form with reference to substituting, thereby comprise individual be similar to (the perhaps center) of K.Because for each audio samples, 6 positions of data are available, so can be for K=64 different being similar to of all different error definition in that part for the specific part of error stream.Can depend on the lossless compress of tabulation then, thereby after compression, finally produce M * 6 bit data samples and N " freedom " 6 bit data sample, wherein L=M+N with L reference.The free space in auxiliary data zone originally can be used to the length of the approximate and synchronous signal mode of memory error, data block, etc.Yet, because can be a series of true random digit at this numerical value that has in the tabulation of L reference, thus should not depend on the compression of tabulation hereto, but guarantee that in fact this tabulation is compressible.Therefore, in the situation of X/Y/Z, and X=24 in this example, Y=I8, Z=6 uses no more than 32=2
(Z-1)Individual approximate.Like this, for reference to this form, only need (Z-1) individual position, and can prove easily that this reference listing is compressible; 5*6 bit data sample can keep form 6 references (5 of each needs) hereto.In 24/18/6 situation,, need 86 data samples to store all data that do not comprise reference listing altogether at least as explaining in the basic fundamental part.(6 (6 bit) samples are used for synchronizing signal; 2 (6 bit) samples are used for data block length, and 2 (6 bit) samples are used for skew, and 6 (6 bit) samples are used for each 2 audio samples copy of 18; 2 (6 bit) is used for decay; With limited 2 (6 bit) data, 64 at the most (6 bit) samples are used for 32 errors and are similar to ... if incompressible, then 2 (6 bit) samples are used for CRC).The given compression ratio (providing 1 free data sample) that is compressed into 5 to 6 of major generals needs 6 * 86=516 sample at the most.This summation also is that this pattern of 24/18/6 defines the maximum data block length.To be similar to numerical limitations and be for example 16, produce 86 to 54 reduction altogether, at least 6 minimum compression ratio and maximum data block lengths of 3 * 54=162 data sample that are compressed into 4 reference listing.Perhaps, through tolerance widths being extended to the 3x6 position, 118 data samples have been produced with all data (this will require 708=6 * 118 altogether) of storage except reference listing.Yet, in most of situations,, be real so further compress the compression of these data because only considered the poorest a kind of situation in the above; For example with 25% (4 are reduced to 3) compression, this is the typical compressed ratio that is used for the approximate form of error.For having approximate being similar to of 32 errors, this especially big ratio can reduce data block length above 50%; 64 the data samples approximate from (32) individual error will be reduced to 48 data samples, thereby total (not containing reference listing) is reduced to 70.Further,, this tabulation is compressed to 5 from 6, further is compressed to 4, thereby produces the data block length that has 3 * 70=210 data sample altogether for the compression of the other 20%-25% of reference listing.As a result, can utilize for the approximate error stream of the approximate reference stream of 32 errors with 210 errors that produce owing to the reduction of audio mixing audio stream sample.
For only having 24/18/6 approximate situation of 16 errors, and take suitable compression ratio, produced the error stream that requires 3 * 46=138 data sample.
At last-based on above instance-but being not limited to the compressed format that these instances-here introduce makes it possible to mode approximate error stream like this; Make when the audio stream that the audio mixing sample frequency is lowered; Can be to this approximate taking in, this will reduce owing to this sample frequency reduces the error that causes significantly.Use that these compressed errors are approximate to be allowed with these two of significant precision reconstruct by the PCM of audio mixing stream, thereby make that to flow the error of introducing be imperceptible to a great extent because combination is with separating these two PCM.
Further requirement, decoder received the information of data block before its handles audio mixing audio samples because it must decoded data block (comprising decompressions) and needs visits these (being similar to) thus the error execution separate audio mixing and operate.Like this, in the phase I of this coding step, A ' also need flow automatically
c' 165a and B '
c' 165b and from second piece of error stream E167 with sample that number is U (=section).Utilization is contained individual 12 the approximate forms of V (=32) and reference listing approximate (using K-intermediate value or facility location algorithm) error stream sample (from the 2nd
nPiece), be associated with the element of that approximate form with each sample with this error stream part.This reference listing constitutes approximate error stream E ' 162.
In combination step<6>In, utilize combiner/formatter audio mixing stream (A '
c', B '
c', E ').This combiner/formatter comprises other amplitude limit analyzer to carry out final inspection about amplitude limit (audio frequency overshoot), and this inspection possibly require small variation.
Combiner/formatter is for example decayed other data; The approximate auxiliary data zone of adding the proper data piece in the data splitting stream that reduces the device generation by sample-sized to of seed numerical value and error; And the output to encoder provides the output stream 169 that comprises mix flow, with the data block portions of audio samples fusion.
The error that reduction will be introduced into through amplitude limit.
Another aspect of the present invention is by preliminary treatment audio stream before the audio mixing effectively.When these signals by audio mixing to a time-out, two or more streams can produce amplitude limit.In this situation, pre-treatment step is on by one of audio mixing passage or even on two passages, comprise the dynamic audio frequency compressor/limiter.This can be through little by little increasing decay before these special events, and those incidents after, little by little reduce and decay and be achieved.This scheme will mainly be applied in the non-fluidisation pattern of encode processor, and this is because it requires (shifting to an earlier date) will produce the sample numerical value of these overshoot/amplitude limits.Can be at audio stream from handling these decay on one's body and therefore avoiding amplitude limit will remain the mode of separating the part that audio mixing flows when these compressor reducer effects when separating audio mixing.Except that the amplitude limit of avoiding (audio mixing) audio frequency, must be to be spendable when (as describing ground in the present invention) to the 2D audio recording when there not being any decoder by the 3D of downward audio mixing.Therefore; On the audio mixing audio stream, use dynamic audio frequency Signal Compression (perhaps decay) to reduce the other audio frequency (from the 3rd dimension) that too much disturbs 2 basic dimension audio frequency; But, recover correct signal level thereby after separating audio mixing, can carry out inverse operation through these attenuation parameters of storage.As stated, the regional block data structure of auxiliary data that forms than low order by sample contains at least 8 the part that is used to keep this dynamic audio frequency compression parameters (decay).In addition, according to analyzing (seeing that sample frequency reduces error correction), can release, about 24/18/6 typical situation of error form with 32 elements and 12 bit error width, the maximum length of data block is 500 samples roughly.Under the sampling rate of 96kHz, this part is about 5 milliseconds audio frequency, and therefore it become the fixed time interval size of attenuation parameter.Decay numerical value self utilizes 8 bit value to represent; When different dB Reduction Levels (for example: 0=0dB is assigned to each numerical value; 1=(0.1) dB; 2=(0.2) dB...) time, can depend on these numerical value and time step to realize level and smooth compression curve, this compression curve can use to recover correct relative signal level with being reversed during decode operation.
Audio stream than low order in storage decay numerical value also can be applied to single stream certainly, the some of them resolution bits is sacrificed in stream, to increase the overall dynamic range of signal in this situation.Alternately; In audio mixing stream; All have relevant decay numerical value thereby can in data block, store a plurality of each data flow of decay numerical value, therefore limit playback level respectively, yet even under low signal level, also keep resolution for each signal for each signal.
In addition; Attenuation parameter can be used for audio mixing 3 dimension audio-frequency informations with mode like this; Make the consumer who does not use these 3 dimension audio-frequency informations can't hear 3 other dimension audio signals; This is because this other signal is attenuated with respect to 2 main dimensional signals, knows that simultaneously the 3 dimensional signal components that this decay numerical value allows the decoder of 3 other dimensional signals of recovery (retrieve) to be attenuated recover (to restore, restore) become its initialize signal level.Usually this required before its audio mixing being become 2 dimension audio frequency PCM streams, for example with 18dB the 3rd dimension audio stream of decaying, flowed with the audio frequency PCM that avoids this audio-frequency information " control " " normally ".This requires other (8) parameter to be defined in the decay on the 3rd dimension audio stream, used before it and another audio stream audio mixing (for each part of audio stream-be defined by length of data block).After through amplification the 3rd dimension audio stream decoding, these 18 decay can be cancelled.
Figure 18 illustrates the AUROPHONIC encoder device.
The input of Aurophonic encoder 184 is a plurality of audio frequency (PCM) passage (voice-grade channels 1 to voice-grade channel X).For each passage, additional position (3D) and the information (position/decay) of the decay when by still less passage of audio mixing one-tenth downwards, used thereof about it.Other input of Aurophonic encoder comprises which audio frequency PCM passage of decision is become the Audio Matrix selection 180 and the Aurophonic encoder character indicating device that is provided for each AURO encoder 181,182,183 of what Aurophonic output channel by downward audio mixing.
The typical input channel of 3D encoder is L (a preceding left side), Lc (preceding LC), C (preceding center), Rc (preceding right center), R (the preceding right side), LFE (low frequency effect), Ls (left side around), Rs (right around), UL (left side of going forward), UC (center of going forward), UR (right side of going forward), ULs (go up a left side around), URs (go up right around), AL (art-left side), AR (art-right side) ....
As being provided by encoder and can reproducing the compatible mutually typical output channel of form with 2D is AURO-L (a left left side) (Aurophonic passage 1); AURO-C (center center) (Aurophonic passage 2); AURO-R (right is right) (Aurophonic passage ...); AURO-Ls (a left surround left side around) (Aurophonic passage ...); AURO-Rs (the right surround right side around) (Aurophonic passage ...); AURO-LFE (Low Freq uency Effects low-frequency effect) (Aurophonic passage Y).
The instance of the AURO coding pass that provides like output by encoder 184:
(AURO-L、AURO-R、AURO-Ls、AURO-Rs)。
AURO-L can comprise initial L (a preceding left side), UL (preceding left side) &AL (art-left side) the pcm audio passage of going up; AURO-R will be a right voice-grade channel before similarly still being used for; AURO-Ls keeps Ls, and (left side is around) &ULs (go up a left side around) audio frequency PCM passage, and AURO-Rs is a right passage of equal value.
Figure 19 illustrates the Aurophonic decoder apparatus.
AUROPHONIC decoder 194 comprises that technology that use is described separates 1 of audio mixing or multitone PCM a plurality of instances 191,192,193 passage, the AURO decoder frequently more in Fig. 5 and 10.For each AURO input channel, AURO decoder 191,192,193 instances are activated.When the AURO passage comprised the audio mixing of 1 voice-grade channel only, the decoder instance should not be activated.
The input of AUROPHONIC decoder receives Aurophonic (PCM) passage Aurophonic passage 1...Aurophonic passage X.For each passage Aurophonic passage 1...Aurophonic passage X, will survey the synchronous signal mode of the AURO data block that whether has the PCM passage automatically as the auxiliary data zone decoder of a part of decoder.When detecting consistent synchronizing signal; AURO decoder 191,192,193 begins to separate the audio-frequency unit of audio mixing AURO (PCM) passage; And (if desired) index that decompresses simultaneously and error form, and with this correct application in separating the audio mixing voice-grade channel.The AURO data comprise that also parameter is like decay (being compensated by decoder) and 3D position.The 3D position is selected to be used in the part 190 to be redirected to the correct output of decoder 194 with separating the audio mixing voice-grade channel in audio frequency output.The user selects the group of audio frequency output channel.
Figure 20 illustrates according to decoder of the present invention.
Since explained all aspects of the present invention, decoder can be able to describe, and comprises advantageous embodiments.
Be used for decoding and preferably automatically survey the technology whether " audio frequency " (for example 24) detailed according to part formerly like the decoder 200 of the signal that obtains by the present invention and be encoded.
This can for example utilize synchronizing signal (synchronously) detector 201 and be implemented, and synchronizing signal detector 201 is searched in received data flow than the synchronous mode in the low order.Synchronizing signal detector 201 has through finding synchronous mode for by the synchronized ability of data block in the auxiliary data zone that forms than low order of sample.Like above explanation ground, to use synchronous mode be alternatively but be favourable.Synchronous signal mode can be that 2,4,6 or 8 (Z-position) is wide and 2,4,6 or 8 samples are long for 24 sample-sized for example.(2 (bit): LSB=01,10; 4: LSB=0001,0010,0100,1000; 6: 000001..., 100000; 8: 00000001..., 10000000).In case synchronizing signal detector 201 has been found that in these match patterns any one, it just " wait " until detecting similar pattern.In case this pattern is detected, synchronizing signal detector 201 just gets into SYNC-candidate-state.Based on the synchronous mode that is detected, synchronizing signal detector 201 can also be confirmed each sample, and whether 2,4,6 or 8 positions are used to the auxiliary data zone.
On the 2nd synchronous signal mode, decoder 200 will scan with decoding block length data block, and utilize next synchronous signal mode check between the starting point of block length and next synchronous signal mode, whether to have coupling.If the two coupling, then decoder 200 gets into the Sync state.If this test crash, then decoder 200 will be always from beginning to restart its synchronous syncing process.During decode operation, decoder 200 is comparison block length and at each number of samples between the initial point of synchronizing signal piece in succession always.In case detected difference, decoder 200 just leaves the Sync state and the syncing process must be restarted.
As in Figure 15 and 16, explaining ground, error correction codes can be applied in data block in the auxiliary data zone to protect existing data.If the form of error correction codes piece is known, and the position of auxiliary data in the error correction codes piece be known, and then this error correction codes can also be used to synchronously.Therefore, in Figure 20, for convenience's sake, synchronizing signal detector and detector for error are shown in combination in the piece 201, but they also can be realized individually.
Detector for error calculates CRC numerical value (using all data from this data block, except synchronizing signal) and this CRC numerical value is compared with the numerical value of finding in the data block end.If there is mismatch, think that then decoder is in the crc error state.
The synchronizing signal detector is to seed numerical value restorer 202; The approximate restorer 203 of error provides information with pilot controller 204, and this information allows seed numerical value restorer 202, the approximate restorer 203 of error and pilot controller 204 from the auxiliary data zone that receives like the input from decoder 200, to extract relevant data.
In case synchronizing signal detector and data block synchronizing signal head are synchronous; Seed numerical value restorer just the data in the scan-data piece to confirm skew, promptly at the terminal of data block and the number of samples between the first authentic copy audio samples (this number can be born in theory) and read these copies (audio frequency) sample.
Seed numerical value restorer 202 recovers one or more seed numerical value and the seed numerical value that returns to is offered separator 206 from the auxiliary data zone of received digital data sets.Separator 206 as explain that at Fig. 5 ground uses the separating basically of seed numerical value (one or more) combine digital data set (unraveling) in 9.The result of this separation or a plurality of digital data sets, or single digital data sets have wherein removed one or more digital data sets from the combined type digital data set.This utilizes three arrows that separator 206 is connected to the output of decoder 200 to illustrate in Figure 20.
Like above explanation ground, use error is approximate to be optional, because the audio frequency that separates like separated device 206 has been very acceptable, and without the approximate error to reduce to introduce through the gradeization of carrying out by encoder of use error.
If desired, the approximate restorer 203 of error is with decompression reference listing and approximate form.If error is approximate will to be used to improve separated digital data sets (one or more), then will be from the error that the approximate restorer 203 of error the receives approximate output that is applied to corresponding digital data set (one or more) and gained digital data sets (one or more) is offered decoder of separator 206.
As long as decoder 200 keeps synchronous with data block head; The approximate restorer 203 of error just will continue decompression reference listing and approximate form, and with these data supplies to separator 206 to separate audio mixing by the audio samples of audio mixing according to C=A "+B "+E ' or C-E '=A "+B ".Separator 206 uses the copy audio samples to become A " sample and B " sample to begin separating audio mixing.For two combined type digital data sets that digital data sets has been combined therein, through adding E '
2i+1Proofread and correct and A '
2iAnd A "
2i+1The A of those even number indexed samples coupling "
2iThe even number indexed samples.Similarly, through adding E '
2i+1Proofread and correct and B '
2i+1And B "
2i+2dThe B of those odd number indexed samples coupling "
2i+1The odd number indexed samples.Go up to use contrary decay at second audio stream (B), and through these samples are zeroized in the least significant bit side to the Z position of shifting left simultaneously, two audio samples (A ' ') all are converted into their initial bit width.Reconstructed sample is issued as dereferenced audio stream independently.
Another selectable unit of decoder 200 is a pilot controller 204.Pilot controller 204 recovers the assist control data that assist control data and processing return to from auxiliary data zone and for example to be used to control self-starter, the form of the control data of musical instrument or lamp provides the result to the auxiliary output of decoder.
In fact; Only the assist control data need be provided at decoder; For example being used for controlling the situation of self-starter corresponding to the mode of the audio stream of combined type digital data centralization, decoder can be by from separator 206, and seed numerical value restorer 202 is removed with the approximate restorer 203 of error.
When decoder got into the CRC-error state, the user can limit the behavior of decoder, and for example he can hope to make the weak one-tenth noise elimination level of second output, in case and decoder recover from its CRC-error state, make that just second output is weak once more.Another kind of behavior can be that audio signal is reappeared two outputs, cause that unfavorable audio frequency falls (plopping) or cracking (cracking) but change at these audio frequency that output place of decoder provides.
Claims (20)
1. one kind is used for the first sample (A with first size
0, A
1, A
2, A
3, A
4, A
5, A
6, A
7, A
8, A
9) the digital audio-frequency data collection (20) and the second sample (B with second size
0, B
1, B
2, B
3, B
4, B
5, B
6, B
7, B
8, B
9) digital audio-frequency data collection (30) is combined into the 3rd sample (C with the 3rd size
0, C
1, C
2, C
3, C
4, C5, C
6, C
7, C
8, C
9) method of digital audio-frequency data collection (40), the 3rd size said method comprising the steps of less than this first size and this second size sum:
-with the first sample subclass (A of the first digital audio-frequency data collection (20)
1, A
3, A
5, A
7, A
9) in each sample etc. change into the second sample subclass (A of the first digital audio-frequency data collection (20)
0, A
2, A
4, A
6, A
8) in adjacent sample, this first sample subclass (A wherein
1, A
3, A
5, A
7, A
9) and this second sample subclass (A
0, A
2, A
4, A
6, A
8) staggered,
-with the 3rd sample subclass (B of the second digital audio-frequency data collection (30)
0, B
2, B
4, B
6, B
8) in each sample etc. change into the 4th sample subclass (B of the second digital audio-frequency data collection (30)
1, B
3, B
5, B
7, B
9) in adjacent sample, the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) staggered, the 4th sample subclass (B wherein
1, B
3, B
5, B
7, B
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) do not have a respective sample any time,
-the sample (A of the first digital audio-frequency data collection through will grade in time domain
0", A
1", A
2", A
3", A
4", A
5", A
6", A
7", A
8", A
9Corresponding sample (the B of ") and the second digital audio-frequency data collection of gradeization
0", B
1", B
2", B
3", B
4", B
5", B
6", B
7", B
8", B
9") addition and produce the sample (C of the 3rd digital audio-frequency data collection
0, C
1, C
2, C
3, C
4, C
5, C
6, C
7, C
8, C
9),
-first seed specimen (the A of this first digital audio-frequency data collection (20) of embedding in the 3rd digital audio-frequency data collection (40)
0) and the second seed specimen (B of this second digital audio-frequency data collection (30)
1).
2. according to the method for claim 1; Wherein, The first digital audio-frequency data collection (20) is represented first audio signal; The second digital audio-frequency data collection (30) is represented second audio signal, and the representative of the 3rd digital audio-frequency data collection (40) is the 3rd audio signal of the combination of first audio signal and second audio signal.
3. according to the method for claim 2; Wherein represent the 4th digital audio-frequency data collection quilt of the 4th audio signal to be combined into the 3rd digital audio-frequency data collection (40) of representing the 3rd audio signal with the first digital audio-frequency data collection (20) and the second digital audio-frequency data collection (30), said the 3rd audio signal is the combination of first audio signal, second audio signal and the 4th audio signal.
4. according to the process of claim 1 wherein that first seed specimen is that first sample and second seed specimen of the first digital audio-frequency data collection is second sample of the second digital audio-frequency data collection.
5. according to the process of claim 1 wherein the first seed specimen (A
0) and the second seed specimen (B
1) be embedded into the sample (C of the 3rd digital audio-frequency data collection (40)
0, C
1, C
2, C
3, C
4, C
5, C
6, C
7, C
8, C
9) than in the low order.
6. according to the process of claim 1 wherein with respect to the first seed specimen (A
0) the position confirmed, position embed synchronous mode (SYNC).
7. according to the process of claim 1 wherein, before the step of sample such as grade, be similar to the caused errors of change such as sample through Select Error from a grouping error is approximate.
8. according to the method for claim 7, wherein, this grouping error is approximate to embed the index of representing this error approximate by index and in the auxiliary data zone (81) that forms than low order through the approximate corresponding sample of this error.
9. one kind is used for from utilizing the 3rd sample (C according to the method acquisition of claim 1
0, C
1, C
2, C
3, C
4, C
5, C
6, C
7, C
8, C
9) digital audio-frequency data collection (40) the extraction first sample (A
0, A
1, A
2, A
3, A
4, A
5, A
6, A
7, A
8, A
9) the digital audio-frequency data collection (20) and the second sample (B
0, B
1, B
2, B
3, B
4, B
5, B
6, B
7, B
8, B
9) method of digital audio-frequency data collection (30), said method comprising the steps of:
-recover the first seed specimen (A of the first digital audio-frequency data collection (20) from the 3rd digital audio-frequency data collection (40)
0) and the second seed specimen (B of the second digital audio-frequency data collection (30)
1),
-recovery comprises the first sample subclass (A
1, A
3, A
5, A
7, A
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) the first digital audio-frequency data collection (20) and comprise the 3rd sample subclass (B
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) the second digital audio-frequency data collection (30), this is the sample (B that extracts the second digital audio-frequency data collection (30) through the known sample numerical value that the respective sample from the 3rd digital audio-frequency data collection (40) deducts the first digital audio-frequency data collection (20)
n), and extract through the known sample numerical value that the respective sample from the 3rd digital audio-frequency data collection (31) deducts the second digital audio-frequency data collection (30) that the sample of the first digital audio-frequency data collection (20) carries out, wherein the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) do not have any time respective sample, wherein first a sample subclass (A
1, A
3, A
5, A
7, A
9) in each sample have and equal the second sample subclass (A
0, A
2, A
4, A
6, A
8) in the numerical value of adjacent sample, the first sample subclass (A wherein
1, A
3, A
5, A
7, A
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) staggered, the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) in each sample have and equal the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) in the numerical value of adjacent sample, and the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) staggered.
10. according to the method for claim 9; Wherein the first digital audio-frequency data collection (20) is represented first audio signal; The second digital audio-frequency data collection (30) is represented second audio signal, and the representative of the 3rd digital audio-frequency data collection (31) is the 3rd audio signal of the combination of first audio signal and second audio signal.
11. method according to claim 10; Wherein represent the 4th digital audio-frequency data collection of the 4th audio signal to be extracted; Said the 4th digital audio-frequency data collection quilt and the first and second digital audio-frequency data collection (20; 30) be combined into the 3rd digital audio-frequency data collection (31) of representing the 3rd audio signal, said the 3rd audio signal is the combination of first audio signal, second audio signal and the 4th audio signal.
12. according to the method for claim 9, wherein first seed specimen is the first sample (A of the first digital audio-frequency data collection
0) and the second seed specimen (B
1) be second sample of the second digital audio-frequency data collection.
13. according to the method for claim 9, the first seed specimen (A wherein
0) and the second seed specimen (B
1) from the sample (C of the 3rd digital audio-frequency data collection (40)
0, C
1, C
2, C
3, C
4, C
5, C
6, C
7, C
8, C
9) extract than low order.
14. according to the method for claim 9, wherein synchronous mode (SYNC) is used to limit the first seed specimen (A
0) the position.
15., wherein, after the step of recovering the first digital audio-frequency data collection, recover the approximate error that during encoding, causes that compensates of error owing to changes such as samples through adding according to the method for claim 9.
16. it is, wherein approximate from recovering error by the auxiliary data zone (81) that forms than low order of the sample of the 3rd digital audio-frequency data collection according to the method for claim 15.
17. an encoder (10) is arranged to carry out the method according to claim 1, comprising:
-the first equalizing apparatus (11a) is used for the first sample subclass (A with the first digital audio-frequency data collection (20)
1, A
3, A
5, A
7, A
9) in each sample etc. change into the second sample subclass (A of the first digital audio-frequency data collection (20)
0, A
2, A
4, A
6, A
8) in adjacent sample, this first sample subclass (A wherein
1, A
3, A
5, A
7, A
9) and this second sample subclass (A
0, A
2, A
4, A
6, A
8) staggered,
-the second equalizing apparatus (11b) is used for the 3rd sample subclass (B with the second digital audio-frequency data collection (30)
0, B
2, B
4, B
6, B
8) in each sample etc. change into the 4th sample subclass (B of the second digital audio-frequency data collection (30)
1, B
3, B
5, B
7, B
9) in adjacent sample, the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) staggered, the 4th sample subclass (B wherein
1, B
3, B
5, B
7, B
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) do not have a respective sample any time,
-combiner (13), be used for through time domain with the sample of the first digital audio-frequency data collection and the corresponding sample addition of the second digital audio-frequency data collection produce the 3rd digital audio-frequency data collection sample and
-formatting mechanism (14) is used for concentrating first seed specimen of this first digital audio-frequency data collection of embedding and second seed specimen of this second digital audio-frequency data collection at the 3rd digital audio-frequency data.
18. a decoder is arranged to carry out the method according to claim 9, comprising:
-seed numerical value restorer (202) is used for from the first seed specimen (A of the 3rd digital audio-frequency data collection (40) the recovery first digital audio-frequency data collection (20)
0) and the second seed specimen (B of the second digital audio-frequency data collection (30)
1),
-processor (206) is used for recovering to comprise the first sample subclass (A
1, A
3, A
5, A
7, A
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) the first digital audio-frequency data collection (20) and comprise the 3rd sample subclass (B
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) the second digital audio-frequency data collection (30), said processor comprises the sample (B that is used to extract the second digital audio-frequency data collection (30)
n) first extractor be used for deducting first subtracter of the known sample numerical value of the first digital audio-frequency data collection (20) from the respective sample of the 3rd digital audio-frequency data collection (40); Said processor further comprises second extractor and second subtracter that is used for deducting from the respective sample of the 3rd digital audio-frequency data collection (31) the known sample numerical value of the second digital audio data set (30), wherein the 4th sample subclass (B of the sample that is used to extract the first digital audio-frequency data collection (20)
1, B
3, B
5, B
7, B
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) do not have any time respective sample, wherein first a sample subclass (A
1, A
3, A
5, A
7, A
9) in each sample have and equal the second sample subclass (A
0, A
2, A
4, A
6, A
8) in the numerical value of adjacent sample, the first sample subclass (A wherein
1, A
3, A
5, A
7, A
9) and the second sample subclass (A
0, A
2, A
4, A
6, A
8) staggered, the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) in each sample have and equal the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) in the numerical value of adjacent sample, and the 3rd sample subclass (B wherein
0, B
2, B
4, B
6, B
8) and the 4th sample subclass (B
1, B
3, B
5, B
7, B
9) staggered, and
Be used to export the output device of the first digital audio-frequency data collection that is resumed.
19. a reproducer comprises the decoder (200) according to claim 18.
20. the vehicle with passenger compartment, said passenger compartment comprises the reproducer according to claim 19, and said reproduction equipment comprises the reader and the amplifier of the data medium that is used to have audio-frequency information.
Applications Claiming Priority (3)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| US82932106P | 2006-10-13 | 2006-10-13 | |
| US60/829,321 | 2006-10-13 | ||
| PCT/EP2007/060980 WO2008043858A1 (en) | 2006-10-13 | 2007-10-15 | A method and encoder for combining digital data sets, a decoding method and decoder for such combined digital data sets and a record carrier for storing such combined digital data set |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN101641970A CN101641970A (en) | 2010-02-03 |
| CN101641970B true CN101641970B (en) | 2012-12-12 |
Family
ID=38983768
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| CN2007800460450A Active CN101641970B (en) | 2006-10-13 | 2007-10-15 | A method and encoder for combining digital data sets, a decoding method and decoder for such combined digital data sets and a record carrier for storing such combined digital data set |
Country Status (13)
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| US (1) | US8620465B2 (en) |
| EP (4) | EP2328364B1 (en) |
| JP (1) | JP5325108B2 (en) |
| CN (1) | CN101641970B (en) |
| AT (1) | ATE476834T1 (en) |
| CA (1) | CA2678681C (en) |
| DE (1) | DE602007008289D1 (en) |
| DK (1) | DK2092791T3 (en) |
| ES (2) | ES2350018T3 (en) |
| HK (1) | HK1141188A1 (en) |
| PL (2) | PL2092791T3 (en) |
| PT (1) | PT2299734E (en) |
| WO (1) | WO2008043858A1 (en) |
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| EP2830052A1 (en) * | 2013-07-22 | 2015-01-28 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Audio decoder, audio encoder, method for providing at least four audio channel signals on the basis of an encoded representation, method for providing an encoded representation on the basis of at least four audio channel signals and computer program using a bandwidth extension |
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Also Published As
| Publication number | Publication date |
|---|---|
| EP2337380B1 (en) | 2020-01-08 |
| EP2092791B1 (en) | 2010-08-04 |
| DE602007008289D1 (en) | 2010-09-16 |
| HK1141188A1 (en) | 2010-10-29 |
| US20100027819A1 (en) | 2010-02-04 |
| CN101641970A (en) | 2010-02-03 |
| EP2337380A1 (en) | 2011-06-22 |
| ES2350018T3 (en) | 2011-01-14 |
| US8620465B2 (en) | 2013-12-31 |
| EP2328364B1 (en) | 2020-07-01 |
| EP2299734A3 (en) | 2011-06-08 |
| WO2008043858A1 (en) | 2008-04-17 |
| ES2399562T3 (en) | 2013-04-02 |
| CA2678681C (en) | 2016-03-22 |
| JP5325108B2 (en) | 2013-10-23 |
| JP2010506226A (en) | 2010-02-25 |
| ATE476834T1 (en) | 2010-08-15 |
| EP2299734B1 (en) | 2012-11-14 |
| EP2337380B8 (en) | 2020-02-26 |
| EP2328364A1 (en) | 2011-06-01 |
| PL2092791T3 (en) | 2011-05-31 |
| PT2299734E (en) | 2013-02-20 |
| PL2299734T3 (en) | 2013-05-31 |
| EP2092791A1 (en) | 2009-08-26 |
| DK2092791T3 (en) | 2010-11-22 |
| EP2299734A2 (en) | 2011-03-23 |
| CA2678681A1 (en) | 2008-04-17 |
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